Re: [Asterisk-biz] Serious Problem with SNOM Attendant phones - Need Help

[C F]

The problem you had with the OP is a know problem (search the wiki)
when used over the network, therfore a good program that uses the
manager app will always use a proxy so that it doesn't communicate
with asterisk over the network. FOP does it this way, and therefore
does'nt have the problem of crashing asterisk.

On 9/28/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> One of my biggest customer, with more than 300 hundred persons using
> asterisk, is having bing problems with the SNOM phones used at the operator
> position (Attendant Console). We currently have 3 operator attending
> incoming calls and bypassing them to the rest of the company.
>
> We did not find any alternative for SNOM to use as Attendant Console (phone
> for the operator). We need Operators to have a keypad where the could
> observe the most important extensions of the office, and be able to transfer
> a call very fast (just pressing a button).
>
> Recently, SNOM released a new model. The SNOM-360 phone, that was ideal for
> that.  However, it only supports one keypad (around 30 extensions),
> althought previous models were able to support several of them. SNOM tolds
> us they are working on this, but they would not have a solution until end of
> the year (at least). We had pay in advance for the models a lot of time ago,
> but it seems there is a lot of problems with SNOM stock, and they did not
> got units to server us. Finally, and due a hard preassure, SNOM sent us
> directly 3 used phones they got in their lab. We received the password of
> only 2 of them.
>
> A week after being in production, we started with a lot of problems with
> them. Such as phones reboot themselfes each 10 minutes. We tryed to upload a
> more recent firmware, but one of them got broken.  We have NOW in production
> just on phone, that still reboots each 10 minutes.  Problem is on the SNOM
> phone, since with all the phones we have connected to Asterisk everything
> goes ok.
>
> Customer is unable to attend all theirs calls, and is really upset.  I need
> to react very fast.
>
> Trying to anticipate to this, one of our developers did a program for the
> Operator. That program used the Asterisk Manager API of asterisk in order to
> check events, and allow the operator to use a program to chek extension
> status and forward the call to any extension very fast. However, it seems
> that Asterisk becomes unstable when has a lot of traffic and your are using
> one of those programas, and having that program installed caused a lot of
> crashes (we think Asterisk Manager API is not very stable, so far).  We do
> not like FOP because it is complicated to use for an untrained operator,
> that could easily lost calls, and do incorrect operations. We need a more
> powerfull software for this.
>
> I saw that 3COM has also attendace phones with a lot of extensions, but they
> do not distribute yet that phone on Europe.  I need a very fast and urgent
> solution for my operators, and have a phone for them asap.
>
> Could any one, please, suggest me any solution ? Any one could serve me
> really fast a Attendance Phone, with a lot of extensions, for using on
> Asterisk to deal with this problem? (please, no more SNOMs, customer does
> not want further to see them).
>
> Since this problem is really urgent I will appreciate if any one having any
> proposal could contact directly with me at andres at europesip.com. Any
> suggestions of specific phones with keypads (to be used as Attendance
> Phones) or programs we could use for the operator, will help a lot.
>
> kind Regards,
>
> Andres
>
>
>
> ___
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> Asterisk-Biz@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>
>
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[Asterisk-biz] (Fwd) [ciscovoip] Special - FCC Mandates VoIP Wiretap Rules -

[Rehan Ahmed AllahWala - Super Technologies I]

--- Forwarded message follows ---
To: [EMAIL PROTECTED]
From:   "talknettalk" <[EMAIL PROTECTED]>
Date sent:  Wed, 28 Sep 2005 19:30:49 -
Subject:[ciscovoip] Special - FCC Mandates VoIP Wiretap Rules - 
Special TECHtionary.com TECH-Tip Und

[ Double-click this line for list subscription options ] 

The animated TECH-Tip tutorial available at 
http://www.techtionary.com

Summary

The FCC has determined that providers of broadband internet access are
not relieved of CALEA obligations.  That is, the FCC has ruled that
broadband internet access classified as an information service under
the Communications Act does not determine its classification for CALEA
purposes.  Up to this point ISPs were considered information services
not communications but already subject to a number of electronic
surveillance statutes that compel their cooperation with law
enforcement agencies.  The ruling also covers "managed voip" services
and rural providers as well.  Implementation of this requirement is
likely in 2007, however, like E911, voip providers are on notice to
get this done.

For the complete FCC Notice http://www.fcc.gov/FCC-05-153A1.pdf

Details

CALEA-Communications Assistance for Law Enforcement Act compliance is
a legal obligation imposed on all carriers covered by the statute. 
According to the FBI, "lawfully-authorized electronic surveillance is
considered to consist of both the interception of communications
content (commonly referred to as wire taps) and the acquisition of
dialing and signaling information used to identify the call (e.g.
DNIS-Dialed Number Information Service) through the use of pen
registers (lawful acquisition of certain outgoing, dialing, routing,
addressing or signaling information) and/or through the use of trap
and trace (source of a wire or electronic communications) devices."


What follows is the standard global legacy interface with the key
interfaces:

HI1-Handover Interface One - Administrative Information

HI2-Handover Interface Two - Intercept Related Information

HI3-Handover Interface Three - Content of Communications

LI1-Lawful Interception interface One - Interception procedures

Shown here is a model for interception of IP-Internet Protocol 
traffic.  Since this part of an ongoing Notice of Proposed 
Rulemaking by the FCC-Federal Communications Commission 
(www.fcc.org), a review of FCC documents is encouraged for the 
latest rules and regulations.  The FCC indicated "the broadband 
access service and ISP-Internet Service Providers may not be able to
easily isolate call-identifying information for VoIP-Voice over
Internet Protocol without examining the packet content."  While it is
understood that nearly every IP is examined in details for data type,
protocol, port, source/destination address, it may be a considerable
effort to retrieve digits dialed or DDE-Dialed Digit Extraction used
in the various methods used in the PSTN-Public Switched Telephone
Network such as inband signaling or SS7-Signaling System 7.

A new standardized approach has emerged for TTP-Trusted Third Party
service called network MS-Mediation Service (L1 Provider Premises) and
LEMF-Law Enforcement Monitoring Facility.  According to the FCC,
"individual carriers are free to choose any technical solution that
meets the assistance capability requirements of CALEA."  There are a
number of different approaches to this issue.   Internal, Adjunct and
Integrated concepts will be discussed next; however there may be many
other viable concepts.  The FCC requires that "all domestic
communicates be transmitted through U.S. earth stations or routes
through a POP-Point of Presence that includes a network switch or
router that is located in the U.S.  While other countries have adapted
CALEA-like regulations, there are many problems to be overcome for
international interception.



Background

The remainder of this tutorial is an introduction not an indepth
analysis of 911.  According to the National Emergency Number
Association, the main characteristic of E-911 is the capability of the
E-911 Tandem CO-Central Office to selectively route a 911 call
originated from any station (telephone number) in the E-911 service
area to the correct primary (controlling) PSAP-Public Service
Answering Point designated to serve the originating station's
location.  

About TECHtionary

TECHtionary.com (http://www.techtionary.com) is the World's First and
Largest Animated Library/Magazine on Technology - Web Hosting
Magazine's Editor's Choice for Technical Help. 

TECHtionary produces white papers, magazine articles, indepth 
product reviews, training tools and custom animations.  






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Post messag

RE: [Asterisk-biz] Places to buy

[The VoIP Connection]

We can help you. -Mike

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED] 

> -Original Message-
> From: Travis Smith [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, September 28, 2005 6:25 PM
> To: asterisk-biz@lists.digium.com
> Subject: [Asterisk-biz] Places to buy
> 
> I am looking for a good place to buy polycom and cisco phones 
> in bulk.any suggestions?
> 
> 
> TS
> 
> 

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[Asterisk-biz] RE Item 2 Travis Smith, Place to buy Polycom and Cisco : Asterisk-Biz Digest, Vol 14, Issue 96

[Shirley TOC]

lance, Expenditure, Payments and number of Calls on each
   account
   - Call Data Records including cost and Sales on each call
   - Branding Module. Allows you to create Brands in any Currency
   - For each Brand define Currency, Billing Increment, mark-up and
   connection charges
   - Flexible Dynamic International Rate Table for Each Brand in any
   Currency
   - Server Status (Displays live status of users phones and ongoing
   calls)
   - Show Peers. List of the last clients(SIP and IAX2) that have
   connected to the Asterisk server.
   - Audit Trail. Show IP, Port and UserAgent for each call
   - Manage your Incoming Public Numbers including Time based forwarding
   - Manage Trunks. You can use unlimited ZAP, IAX and SIP trunks.
   - Time Based Trunk Dialing. Each trunk can have his own time based
   dialplan
   - Temporary disable trunks
   - Trunks can be rated after cost. Allow for cost based Dialing
   - Define maximum concurrent outgoing calls on each trunk
   - If lowest cost trunk is fully used (busy) the system will choose the
   next available trunk.
   - Define unlimited outgoing routes and link them to your trunks
   - Store cost of your outgoing route for each trunk for efficient cost
   control
   - All outgoing routes are stored independent on the client price list
   - Define customer price lists for each Brand and Currency
   - Advanced customer management andportal management
   - Integrated E-commerce module and web shop is available under GPL
   - Define list of VOIP hardware commonly used
   - Full Hardware Inventory. Store mac address and serial numbers of
   client hardware
   - View and Store Customers payments
   - Asterisk Billing and Management
   - Manage Pre Paid and Post Paid customers. Full Credit control by User
   Account
   - View important Server logs from web interface
   - Define maximum concurrent calls on each Customer Account

AstBill includes the functionality needed by most small- to medium-sized
businesses (SMB). It is also very efficient as a platform for small VOIP
providers. AstBill is used by several small businesses and is the platform
for http://astartelecom.com the new European Asterisk call termination
service currently in beta.

Are Casilla
-- next part --
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--

Message: 2
Date: Wed, 28 Sep 2005 17:24:40 -0500
From: Travis Smith <[EMAIL PROTECTED]>
Subject: [Asterisk-biz] Places to buy
To: asterisk-biz@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

I am looking for a good place to buy polycom and cisco phones in
bulk.any suggestions?


TS


--

Message: 3
Date: Wed, 28 Sep 2005 18:31:34 -0400
From: "Billy Huddleston" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-biz] Multitenant PBX configuration
To: "Commercial and Business-Oriented Asterisk Discussion"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original

Does it support realtime SIP & Extensions?

- Original Message -
From: "Alex Epshteyn" <[EMAIL PROTECTED]>
To: "'Commercial and Business-Oriented Asterisk Discussion'"

Sent: Wednesday, September 28, 2005 5:39 PM
Subject: RE: [Asterisk-biz] Multitenant PBX configuration


> Oops, username is demo, password insecure
>
>
>
>  _
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brian
> Franklin -
> NTG
> Sent: Wednesday, September 28, 2005 2:34 PM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [Asterisk-biz] Multitenant PBX configuration
>
>
>
> What is the DEMO username and password?
>
> - Original Message -
>
> From: Jonathan k. <mailto:[EMAIL PROTECTED]>  Creasy
>
> To: Commercial and Business-Oriented Asterisk
> <mailto:asterisk-biz@lists.digium.com>  Discussion
>
> Sent: Wednesday, September 28, 2005 4:08 PM
>
> Subject: RE: [Asterisk-biz] Multitenant PBX configuration
>
>
>
>
>
>
>
>
>  _
>
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn
> Sent: Wednesday, September 28, 2005 5:01 PM
> To: 'Commercial and Business-Oriented Asterisk Discussion'
> Subject: [Asterisk-biz] Multitenant PBX configuration
>
>
>
> We are adding multi tenant capabilities to Asterisk PBX Manager
> http://www.thirdlane.com/opensource.htm#manager, and would really
> appreciate
> input from the list. Here are a few questions.
>
>
>
> 1) Would you find it useful to have multi tenant system with file
> based
> configuration (which is 

Re: [Asterisk-biz] Multitenant PBX configuration

[Billy Huddleston]

How soon? And, do we get the source code for the application (So that we can 
customize it if needed...)


Thanks, Billy

- Original Message - 
From: "Alex Epshteyn" <[EMAIL PROTECTED]>
To: "'Commercial and Business-Oriented Asterisk Discussion'" 


Sent: Wednesday, September 28, 2005 6:43 PM
Subject: RE: [Asterisk-biz] Multitenant PBX configuration



Not currently, we will be adding that soon.


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-biz-
[EMAIL PROTECTED] On Behalf Of Billy Huddleston
Sent: Wednesday, September 28, 2005 3:32 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Multitenant PBX configuration

Does it support realtime SIP & Extensions?

- Original Message -
From: "Alex Epshteyn" <[EMAIL PROTECTED]>
To: "'Commercial and Business-Oriented Asterisk Discussion'"

Sent: Wednesday, September 28, 2005 5:39 PM
Subject: RE: [Asterisk-biz] Multitenant PBX configuration


> Oops, username is demo, password insecure
>
>
>
>  _
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Brian
> Franklin -
> NTG
> Sent: Wednesday, September 28, 2005 2:34 PM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [Asterisk-biz] Multitenant PBX configuration
>
>
>
> What is the DEMO username and password?
>
> - Original Message -
>
> From: Jonathan k.   Creasy
>
> To: Commercial and Business-Oriented Asterisk
>   Discussion
>
> Sent: Wednesday, September 28, 2005 4:08 PM
>
> Subject: RE: [Asterisk-biz] Multitenant PBX configuration
>
>
>
>
>
>
>
>
>  _
>
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Alex
Epshteyn
> Sent: Wednesday, September 28, 2005 5:01 PM
> To: 'Commercial and Business-Oriented Asterisk Discussion'
> Subject: [Asterisk-biz] Multitenant PBX configuration
>
>
>
> We are adding multi tenant capabilities to Asterisk PBX Manager
> http://www.thirdlane.com/opensource.htm#manager, and would really
> appreciate
> input from the list. Here are a few questions.
>
>
>
> 1) Would you find it useful to have multi tenant system with file
> based
> configuration (which is what PBX Manager is currently) or database
> /realtime
> configuration would always be required for your typical multi tenant
> setup?
>
>
>
> 2) Would you prefer the system where all the management be 
> performed

> by
> one "super admin" or you would prefer that each tenant would have their
> own
> administrator with a separate login, etc. Would you prefer to have 
> both,

> in
> which case which functions would you allow for the tenant 
> administrator?

>
>
>
> 3) Assuming that the tenant has to be selected first, would you
prefer
> the GUI display only the tenant's extensions, phones, maulboxes, etc, 
> or

> all
> of these as well as a filed showing which tenant they belong to? Note
that
> all the lists are displayed when selection arguments in scripts 
> assigned

> to
> extensions or menu items, so showing everything may be too much.
>
>
>
> 4) Should outbound trunks be defined for each tenant or shared?
>
>
>
> 5) Would a simple convention of prepending tenant name to phones,
> voice
> menus, queues, voicemail contexts, etc work for you?
>
>
>
> 6) Would you prefer to share the music on hold (where various
classes
> and playlists could be defined and assigned to any tenant) but not
> strictly
> separated by the GUI or have a strict separation (when each class can
only
> be used by one tenant).
>
>
>
> 7) What about sharing conference rooms?
>
>
>
>
>
> You can look at the demo of our current software at
> https://thirdlane.amer.net:1/asterisk, Any other requests or
> suggestions, for the multi tenant and single tenant versions of 
> software

> are
> welcome.
>
>
>
> Thanks,
>
>
>
> Alex
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>  _
>
>
> ___
> Asterisk-Biz mailing list
> Asterisk-Biz@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>
>  _
>
>
> No virus found in this incoming message.
> Checked by AVG Anti-Virus.
> Version: 7.0.344 / Virus Database: 267.11.8/113 - Release Date:
9/27/2005
>
>


--
--


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>

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Re: [Asterisk-biz] Places to buy

[Ariel Batista]

Voipsupply.com
On 9/28/05, Travis Smith <[EMAIL PROTECTED]> wrote:
I am looking for a good place to buy polycom and cisco phones inbulk.any suggestions?TS
___Asterisk-Biz mailing listAsterisk-Biz@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-biz-- Ariel BatistaKasi InternationalOffice: 305-574-6721  FX: 305-574-0212Home: 305-574-1495 Cell: 786-376-1896 
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Re: [Asterisk-biz] Places to buy

[BJ Weschke]

voipsupply.com ? 
 
they're the only place I would go to do so. 
On 9/28/05, Travis Smith <[EMAIL PROTECTED]> wrote:
I am looking for a good place to buy polycom and cisco phones inbulk.any suggestions?TS
___Asterisk-Biz mailing listAsterisk-Biz@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-biz
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RE: [Asterisk-biz] Multitenant PBX configuration

[Alex Epshteyn]

Not currently, we will be adding that soon.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-biz-
> [EMAIL PROTECTED] On Behalf Of Billy Huddleston
> Sent: Wednesday, September 28, 2005 3:32 PM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [Asterisk-biz] Multitenant PBX configuration
> 
> Does it support realtime SIP & Extensions?
> 
> - Original Message -
> From: "Alex Epshteyn" <[EMAIL PROTECTED]>
> To: "'Commercial and Business-Oriented Asterisk Discussion'"
> 
> Sent: Wednesday, September 28, 2005 5:39 PM
> Subject: RE: [Asterisk-biz] Multitenant PBX configuration
> 
> 
> > Oops, username is demo, password insecure
> >
> >
> >
> >  _
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Brian
> > Franklin -
> > NTG
> > Sent: Wednesday, September 28, 2005 2:34 PM
> > To: Commercial and Business-Oriented Asterisk Discussion
> > Subject: Re: [Asterisk-biz] Multitenant PBX configuration
> >
> >
> >
> > What is the DEMO username and password?
> >
> > - Original Message -
> >
> > From: Jonathan k.   Creasy
> >
> > To: Commercial and Business-Oriented Asterisk
> >   Discussion
> >
> > Sent: Wednesday, September 28, 2005 4:08 PM
> >
> > Subject: RE: [Asterisk-biz] Multitenant PBX configuration
> >
> >
> >
> >
> >
> >
> >
> >
> >  _
> >
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Alex
> Epshteyn
> > Sent: Wednesday, September 28, 2005 5:01 PM
> > To: 'Commercial and Business-Oriented Asterisk Discussion'
> > Subject: [Asterisk-biz] Multitenant PBX configuration
> >
> >
> >
> > We are adding multi tenant capabilities to Asterisk PBX Manager
> > http://www.thirdlane.com/opensource.htm#manager, and would really
> > appreciate
> > input from the list. Here are a few questions.
> >
> >
> >
> > 1) Would you find it useful to have multi tenant system with file
> > based
> > configuration (which is what PBX Manager is currently) or database
> > /realtime
> > configuration would always be required for your typical multi tenant
> > setup?
> >
> >
> >
> > 2) Would you prefer the system where all the management be performed
> > by
> > one "super admin" or you would prefer that each tenant would have their
> > own
> > administrator with a separate login, etc. Would you prefer to have both,
> > in
> > which case which functions would you allow for the tenant administrator?
> >
> >
> >
> > 3) Assuming that the tenant has to be selected first, would you
> prefer
> > the GUI display only the tenant's extensions, phones, maulboxes, etc, or
> > all
> > of these as well as a filed showing which tenant they belong to? Note
> that
> > all the lists are displayed when selection arguments in scripts assigned
> > to
> > extensions or menu items, so showing everything may be too much.
> >
> >
> >
> > 4) Should outbound trunks be defined for each tenant or shared?
> >
> >
> >
> > 5) Would a simple convention of prepending tenant name to phones,
> > voice
> > menus, queues, voicemail contexts, etc work for you?
> >
> >
> >
> > 6) Would you prefer to share the music on hold (where various
> classes
> > and playlists could be defined and assigned to any tenant) but not
> > strictly
> > separated by the GUI or have a strict separation (when each class can
> only
> > be used by one tenant).
> >
> >
> >
> > 7) What about sharing conference rooms?
> >
> >
> >
> >
> >
> > You can look at the demo of our current software at
> > https://thirdlane.amer.net:1/asterisk, Any other requests or
> > suggestions, for the multi tenant and single tenant versions of software
> > are
> > welcome.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Alex
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >  _
> >
> >
> > ___
> > Asterisk-Biz mailing list
> > Asterisk-Biz@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-biz
> >
> >
> >  _
> >
> >
> > No virus found in this incoming message.
> > Checked by AVG Anti-Virus.
> > Version: 7.0.344 / Virus Database: 267.11.8/113 - Release Date:
> 9/27/2005
> >
> >
> 
> 
> --
> --
> 
> 
> > ___
> > Asterisk-Biz mailing list
> > Asterisk-Biz@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-biz
> >
> 
> ___
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Re: [Asterisk-biz] AstBil New Billing, Routing and Management software for Asterisk

[Operations Jabeens]

Dear Are,
 
I am developing a business (home based) for a targeted retail audience services of calling card (IVR based), ATA, PC2 Phone (Dialer), ANI (both PSTN and IMSI Mobile), web call, call shoo, wholesale (post paid) and SMS call.

 
It will be introduced to immigrant community of Bangladesh with primary channel of sales being commerce site along physical distribution of card and reseller program.
 
Your announcement has struck me that if your are interested you can work as a paid consultant in this project. Please let me know if this interest you. 
 
Thanks,
Ifte Islam
Jabeens Telecom LLC, USA
+1 301 792 3849
MSN: [EMAIL PROTECTED]
 
 
On 9/28/05, London3 <[EMAIL PROTECTED]> wrote:

AstBill is not only a web-based, user-friendly billing interface to Asterisk. It is also a Asterisk configuration and management tool and a standardized implementation of Asterisk using REALTIME and static configuration as you please.

http://astbill.com
AstBill DEMO: http://demo.astbill.com
Here are some of the features of AstBill:
User-friendly End User Web interface gives access to a range of functionality:

Personal Contact Directory with Categories 
View SIP, IAX and Virtual Accounts 
Virtual Accounts (Lets you forward your calls to any extension you want also time based forwarding) 
Credit Control on outgoing calls 
Show Balance, Expenditure, Payments and number of Calls on each account 
Set warning balance for email when low credit on account 
View Numbers Dialed and add them to the Contact Directory 
View Numbers Dialed by Names from the Contact Directory 
Dynamic International Rate Table(Each customer can have his own price list using Brands) 
Rate Table in his Currency of choice 
Call Data Records including cost of each call and time based billing 
Call Data Records in his Currency of choice 
Switchboard (Displays live status of users phones and ongoing calls) 
Allows one click calling from GUI and direct to phone 
Call Parking sends calls to parking and then redirects to phone 
Allows transfers of calls 
Edit your Account setup 
Asterisk Billing and Management 
Edit voicemail setup including email and pin 
Create Time Based Dialing. Allows you to forward your calls based on time and day. 
Each user can have unlimited of SIP, IAX and Virtual Accounts 
Each user can have unlimited Prepaid Card Accounts linked to his userid 
Specify your hardware and change the viewable name of your accounts 
Temporary disable SIP, IAX or Virtual Account 
Manage your Incoming Public Numbers including Time based forwarding 
User-friendly Administrator Web interface gives access to a range of functionality:

Show Balance, Expenditure, Payments and number of Calls on each account 
Call Data Records including cost and Sales on each call 
Branding Module. Allows you to create Brands in any Currency 
For each Brand define Currency, Billing Increment, mark-up and connection charges 
Flexible Dynamic International Rate Table for Each Brand in any Currency 
Server Status (Displays live status of users phones and ongoing calls) 
Show Peers. List of the last clients(SIP and IAX2) that have connected to the Asterisk server. 
Audit Trail. Show IP, Port and UserAgent for each call 
Manage your Incoming Public Numbers including Time based forwarding 
Manage Trunks. You can use unlimited ZAP, IAX and SIP trunks. 
Time Based Trunk Dialing. Each trunk can have his own time based dialplan 
Temporary disable trunks 
Trunks can be rated after cost. Allow for cost based Dialing 
Define maximum concurrent outgoing calls on each trunk 
If lowest cost trunk is fully used (busy) the system will choose the next available trunk. 
Define unlimited outgoing routes and link them to your trunks 
Store cost of your outgoing route for each trunk for efficient cost control 
All outgoing routes are stored independent on the client price list 
Define customer price lists for each Brand and Currency 
Advanced customer management andportal management 
Integrated E-commerce module and web shop is available under GPL 
Define list of VOIP hardware commonly used 
Full Hardware Inventory. Store mac address and serial numbers of client hardware 
View and Store Customers payments 
Asterisk Billing and Management 
Manage Pre Paid and Post Paid customers. Full Credit control by User Account 
View important Server logs from web interface 
Define maximum concurrent calls on each Customer Account 
AstBill includes the functionality needed by most small- to medium-sized businesses (SMB). It is also very efficient as a platform for small VOIP providers. AstBill is used by several small businesses and is the platform for 
http://astartelecom.com the new European Asterisk call termination service currently in beta.
Are Casilla___Asterisk-Biz mailing listAsterisk-Biz@lists.digium.com
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Re: [Asterisk-biz] Multitenant PBX configuration

[Billy Huddleston]

Does it support realtime SIP & Extensions?

- Original Message - 
From: "Alex Epshteyn" <[EMAIL PROTECTED]>
To: "'Commercial and Business-Oriented Asterisk Discussion'" 


Sent: Wednesday, September 28, 2005 5:39 PM
Subject: RE: [Asterisk-biz] Multitenant PBX configuration



Oops, username is demo, password insecure



 _

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian 
Franklin -

NTG
Sent: Wednesday, September 28, 2005 2:34 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Multitenant PBX configuration



What is the DEMO username and password?

- Original Message - 


From: Jonathan k.   Creasy

To: Commercial and Business-Oriented Asterisk
  Discussion

Sent: Wednesday, September 28, 2005 4:08 PM

Subject: RE: [Asterisk-biz] Multitenant PBX configuration








 _


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn
Sent: Wednesday, September 28, 2005 5:01 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz] Multitenant PBX configuration



We are adding multi tenant capabilities to Asterisk PBX Manager
http://www.thirdlane.com/opensource.htm#manager, and would really 
appreciate

input from the list. Here are a few questions.



1) Would you find it useful to have multi tenant system with file 
based
configuration (which is what PBX Manager is currently) or database 
/realtime
configuration would always be required for your typical multi tenant 
setup?




2) Would you prefer the system where all the management be performed 
by
one "super admin" or you would prefer that each tenant would have their 
own
administrator with a separate login, etc. Would you prefer to have both, 
in

which case which functions would you allow for the tenant administrator?



3) Assuming that the tenant has to be selected first, would you prefer
the GUI display only the tenant's extensions, phones, maulboxes, etc, or 
all

of these as well as a filed showing which tenant they belong to? Note that
all the lists are displayed when selection arguments in scripts assigned 
to

extensions or menu items, so showing everything may be too much.



4) Should outbound trunks be defined for each tenant or shared?



5) Would a simple convention of prepending tenant name to phones, 
voice

menus, queues, voicemail contexts, etc work for you?



6) Would you prefer to share the music on hold (where various classes
and playlists could be defined and assigned to any tenant) but not 
strictly

separated by the GUI or have a strict separation (when each class can only
be used by one tenant).



7) What about sharing conference rooms?





You can look at the demo of our current software at
https://thirdlane.amer.net:1/asterisk, Any other requests or
suggestions, for the multi tenant and single tenant versions of software 
are

welcome.



Thanks,



Alex














 _


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[Asterisk-biz] Places to buy

[Travis Smith]

I am looking for a good place to buy polycom and cisco phones in 
bulk.any suggestions?



TS
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[Asterisk-biz] AstBil New Billing, Routing and Management software for Asterisk

[London3]

AstBill is not only a web-based, user-friendly billing interface to
Asterisk. It is also a Asterisk configuration and management tool and a
standardized implementation of Asterisk using REALTIME and static
configuration as you please.

http://astbill.com
AstBill DEMO: http://demo.astbill.com

Here are some of the features of AstBill:

User-friendly End User Web interface gives access to a range of functionality:



Personal Contact Directory with CategoriesView SIP, IAX and Virtual AccountsVirtual Accounts (Lets you forward your calls to any extension you want also time based forwarding)Credit Control on outgoing calls 
Show Balance, Expenditure, Payments and number of Calls on each accountSet warning balance for email when low credit on accountView Numbers Dialed and add them to the Contact Directory 
View Numbers Dialed by Names from the Contact Directory Dynamic International Rate Table(Each customer can have his own price list using Brands)Rate Table in his Currency of choice Call Data Records including cost of each call and time based billing 
Call Data Records in his Currency of choice Switchboard (Displays live status of users phones and ongoing calls) Allows one click calling from GUI and direct to phone Call Parking sends calls to parking and then redirects to phone 
Allows transfers of calls Edit your Account setup Asterisk Billing and ManagementEdit voicemail setup including email and pin Create Time Based Dialing. Allows you to forward your calls based on time and day. 
Each user can have unlimited of SIP, IAX and Virtual Accounts Each user can have unlimited Prepaid Card Accounts linked to his userid Specify your hardware and change the viewable name of your accounts 
Temporary disable SIP, IAX or Virtual Account Manage your Incoming Public Numbers including Time based forwarding 



User-friendly Administrator Web interface gives access to a range of functionality:



Show Balance, Expenditure, Payments and number of Calls on each account Call Data Records including cost and Sales on each call Branding Module. Allows you to create Brands in any Currency 
For each Brand define Currency, Billing Increment, mark-up and connection charges Flexible Dynamic International Rate Table for Each Brand in any Currency Server Status (Displays live status of users phones and ongoing calls) 
Show Peers. List of the last clients(SIP and IAX2) that have connected to the Asterisk server. Audit Trail. Show IP, Port and UserAgent for each call Manage your Incoming Public	Numbers including Time based forwarding 
Manage Trunks. You can use unlimited ZAP, IAX and SIP trunks. Time Based Trunk Dialing. Each trunk can have his own time based dialplan Temporary disable trunks Trunks can be rated after cost. Allow for cost based Dialing 
Define maximum concurrent outgoing calls on each trunk If lowest cost trunk is fully used (busy) the system will choose the next available trunk. Define unlimited outgoing routes and link them to your trunks 
Store cost of your outgoing route for each trunk for efficient cost control All outgoing routes are stored independent on the client price list Define customer price lists for each Brand and Currency 
Advanced customer management andportal managementIntegrated E-commerce module and web shop is available under GPL Define list of VOIP hardware commonly used Full Hardware Inventory. Store mac address and serial numbers of client hardware 
View and Store Customers payments Asterisk Billing and ManagementManage Pre Paid and Post Paid customers. Full Credit control by User Account View important Server logs from web interface 
Define maximum concurrent calls on each Customer Account 



AstBill includes the functionality needed by most small- to
medium-sized businesses (SMB). It is also very efficient as a platform
for small VOIP providers. AstBill is used by several small businesses
and is the platform for http://astartelecom.com the new European Asterisk call termination service currently in beta.

Are Casilla

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RE: [Asterisk-biz] Multitenant PBX configuration

[Alex Epshteyn]

Oops, username is demo, password insecure

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Franklin - NTG
Sent: Wednesday, September 28,
2005 2:34 PM
To: Commercial and
Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz]
Multitenant PBX configuration



 



What is the DEMO username and password?







- Original Message - 





From: Jonathan k.
Creasy 





To: Commercial and Business-Oriented Asterisk
Discussion 





Sent: Wednesday,
September 28, 2005 4:08 PM





Subject: RE: [Asterisk-biz]
Multitenant PBX configuration





 



 

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn
Sent: Wednesday, September 28,
2005 5:01 PM
To: 'Commercial and
Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz]
Multitenant PBX configuration



 

We are adding multi tenant capabilities to
Asterisk PBX Manager http://www.thirdlane.com/opensource.htm#manager, and would really appreciate input from
the list. Here are a few questions.

 

1) Would you find it useful to have multi tenant system with file
based configuration (which is what PBX Manager is currently) or database
/realtime configuration would always be required for your typical multi tenant
setup?

 

2) Would you prefer the system where all the management be performed
by one “super admin” or you would prefer that each tenant would
have their own administrator with a separate login, etc. Would you prefer to
have both, in which case which functions would you allow for the tenant
administrator?

 

3) Assuming that the tenant has to be selected first, would you prefer
the GUI display only the tenant’s extensions, phones, maulboxes, etc, or
all of these as well as a filed showing which tenant they belong to? Note that
all the lists are displayed when selection arguments in scripts assigned to
extensions or menu items, so showing everything may be too much.

 

4) Should outbound trunks be defined for each tenant or shared?

 

5) Would a simple convention of prepending tenant name to phones,
voice menus, queues, voicemail contexts, etc work for you?

 

6) Would you prefer to share the music on hold (where various classes
and playlists could be defined and assigned to any tenant) but not strictly
separated by the GUI or have a strict separation (when each class can only be
used by one tenant). 

 

7) What about sharing conference rooms?

 

 

You can look at the demo of our current
software at https://thirdlane.amer.net:1/asterisk,
Any other requests or suggestions, for the multi
tenant and single tenant versions of software are welcome.

 

Thanks,

 

Alex

 

 

 

 

 

 







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Re: [Asterisk-biz] Canada - DIDs

[Rehan Ahmed AllahWala - Super Technologies I]

Dear Sir,

You can request them on www.didx.org

Rehan



>
> Hello,
>
> I’m looking for DIDs for the following city in Canada: Quebec City,
> Sherbrooke and I’m also interested about any other DIDs in any other
> city in Canada.
>
> Thanks for you time and you help.
>
> Best regards,
>
> ==
> Pierre-Luc Quimper
> GloboTech Communications
> Phone: 1-514-907-0050
> Toll Free: 1-(888)-GTCOMM1
> Phone Urgency: 1-(506) 235-9182
>  1-(866)-802-2200
> Fax: 1-(506) 235-2980
> [EMAIL PROTECTED]
> http://www.gtcomm.net

Super Technologies Inc., Pensacola, Florida
http://www.supertec.com - Technologies from tomorrow, TODAY!
http://www.VirtualPhoneLine.com - Get A US, UK, EU Number, Forward it to
any  number in the world instantly.
http://www.PhoneOpia.com - SIP Based OPEN Phone Services.
http://www.MySuperPhone.com - The NEXT Generation of Telephone.
Http://www.ip-pabx.com - One World, One Number, One Pabx, All Hosted.
Http://www.superPBX.net - One World, One Number, One Pabx, Physical.
http://www.didX.org - We will sell all your DID stock for you.


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Re: [Asterisk-biz] Junction Networks, SellVoIP or IAX.CC?

[Rehan Ahmed AllahWala - Super Technologies I]

Dear Carlos, look at didx.org if you are a service provider

Rehan



> 
> Hello!
> 
> I'm looking for IAX origination/termination for my small business with
> the following features:
> 
> 1) At least 4 IAX incoming/outgoing simultaneous calls
> 
> 2) Availability of Vanity toll-free numbers
> 
> 3) Availability of DID in Herndon VA (area code 703)
> 
> 4) Doesn't break the bank
> 
> So far, I've look at Junction Networks, SellVoIP and IAX.CC. Anything
> good/bad to say about these? Any others I should be aware of?
> 
> Thanks!
> 
> Carlos
> -- 
> "We hold [...] that all men are created equal; that they are
> endowed [...] with certain inalienable rights; that among
> these are life, liberty, and the pursuit of happiness"
> -- Thomas Jefferson 

Super Technologies Inc., Pensacola, Florida
http://www.supertec.com - Technologies from tomorrow, TODAY!
http://www.VirtualPhoneLine.com - Get A US, UK, EU Number, Forward it to
any  number in the world instantly.
http://www.PhoneOpia.com - SIP Based OPEN Phone Services.
http://www.MySuperPhone.com - The NEXT Generation of Telephone.
Http://www.ip-pabx.com - One World, One Number, One Pabx, All Hosted.
Http://www.superPBX.net - One World, One Number, One Pabx, Physical.
http://www.didX.org - We will sell all your DID stock for you.


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Re: [Asterisk-biz] Multitenant PBX configuration

[Brian Franklin - NTG]

What is the DEMO username and 
password?

  - Original Message - 
  From: 
  Jonathan 
  k. Creasy 
  To: Commercial and Business-Oriented 
  Asterisk Discussion 
  Sent: Wednesday, September 28, 2005 4:08 
  PM
  Subject: RE: [Asterisk-biz] Multitenant 
  PBX configuration
  
  
   
   
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Alex EpshteynSent: Wednesday, September 28, 2005 5:01 
  PMTo: 'Commercial and 
  Business-Oriented Asterisk Discussion'Subject: [Asterisk-biz] Multitenant PBX 
  configuration
   
  We are adding multi 
  tenant capabilities to Asterisk PBX Manager http://www.thirdlane.com/opensource.htm#manager, and would really appreciate input from 
  the list. Here are a few questions.
   
  1)   
  Would 
  you find it useful to have multi tenant system with file based configuration 
  (which is what PBX Manager is currently) or database /realtime configuration 
  would always be required for your typical multi tenant 
  setup?
   
  2)   
  Would 
  you prefer the system where all the management be performed by one “super 
  admin” or you would prefer that each tenant would have their own administrator 
  with a separate login, etc. Would you prefer to have both, in which case which 
  functions would you allow for the tenant 
  administrator?
   
  3)   
  Assuming 
  that the tenant has to be selected first, would you prefer the GUI display 
  only the tenant’s extensions, phones, maulboxes, etc, or all of these as well 
  as a filed showing which tenant they belong to? Note that all the lists are 
  displayed when selection arguments in scripts assigned to extensions or menu 
  items, so showing everything may be too much.
   
  4)   
  Should 
  outbound trunks be defined for each tenant or 
  shared?
   
  5)   
  Would a 
  simple convention of prepending tenant name to phones, voice menus, queues, 
  voicemail contexts, etc work for you?
   
  6)   
  Would 
  you prefer to share the music on hold (where various classes and playlists 
  could be defined and assigned to any tenant) but not strictly separated by the 
  GUI or have a strict separation (when each class can only be used by one 
  tenant). 
   
  7)   
  What 
  about sharing conference rooms?
   
   
  You can look at the 
  demo of our current software at https://thirdlane.amer.net:1/asterisk, 
  Any other requests or 
  suggestions, for the multi tenant and single tenant versions of software are 
  welcome.
   
  Thanks,
   
  Alex
   
   
   
   
   
   
  
  

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  listAsterisk-Biz@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-biz
  
  

  No virus found in this incoming message.Checked by AVG 
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  9/27/2005
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RE: [Asterisk-biz] Multitenant PBX configuration

[Jonathan k. Creasy]

 

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn
Sent: Wednesday, September 28,
2005 5:01 PM
To: 'Commercial and
Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz]
Multitenant PBX configuration



 

We are adding multi tenant capabilities to
Asterisk PBX Manager http://www.thirdlane.com/opensource.htm#manager, and would really appreciate input from
the list. Here are a few questions.

 

1)   Would you find it useful to have multi tenant system with file
based configuration (which is what PBX Manager is currently) or database
/realtime configuration would always be required for your typical multi tenant
setup?

 

2)   Would you prefer the system where all the management be performed
by one “super admin” or you would prefer that each tenant would
have their own administrator with a separate login, etc. Would you prefer to
have both, in which case which functions would you allow for the tenant
administrator?

 

3)   Assuming that the tenant has to be selected first, would you prefer
the GUI display only the tenant’s extensions, phones, maulboxes, etc, or
all of these as well as a filed showing which tenant they belong to? Note that
all the lists are displayed when selection arguments in scripts assigned to
extensions or menu items, so showing everything may be too much.

 

4)   Should outbound trunks be defined for each tenant or shared?

 

5)   Would a simple convention of prepending tenant name to phones,
voice menus, queues, voicemail contexts, etc work for you?

 

6)   Would you prefer to share the music on hold (where various classes
and playlists could be defined and assigned to any tenant) but not strictly
separated by the GUI or have a strict separation (when each class can only be
used by one tenant). 

 

7)   What about sharing conference rooms?

 

 

You can look at the demo of our current
software at https://thirdlane.amer.net:1/asterisk,
Any other requests or suggestions, for the multi
tenant and single tenant versions of software are welcome.

 

Thanks,

 

Alex

 

 

 

 

 

 






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[Asterisk-biz] Multitenant PBX configuration

[Alex Epshteyn]

We are adding multi tenant capabilities to
Asterisk PBX Manager http://www.thirdlane.com/opensource.htm#manager, and would really appreciate input from
the list. Here are a few questions.

 

1) Would you find it useful to have multi tenant system with file
based configuration (which is what PBX Manager is currently) or database /realtime
configuration would always be required for your typical multi tenant setup?

 

2) Would you prefer the system where all the management be performed
by one “super admin” or you would prefer that each tenant would
have their own administrator with a separate login, etc. Would you prefer to
have both, in which case which functions would you allow for the tenant
administrator?

 

3) Assuming that the tenant has to be selected first, would you prefer
the GUI display only the tenant’s extensions, phones, maulboxes, etc, or
all of these as well as a filed showing which tenant they belong to? Note that
all the lists are displayed when selection arguments in scripts assigned to
extensions or menu items, so showing everything may be too much.

 

4) Should outbound trunks be defined for each tenant or shared?

 

5) Would a simple convention of prepending tenant name to phones,
voice menus, queues, voicemail contexts, etc work for you?

 

6) Would you prefer to share the music on hold (where various classes
and playlists could be defined and assigned to any tenant) but not strictly separated
by the GUI or have a strict separation (when each class can only be used by one
tenant). 

 

7) What about sharing conference rooms?

 

 

You can look at the demo of our current software
at https://thirdlane.amer.net:1/asterisk,
Any other requests or suggestions, for the multi
tenant and single tenant versions of software are welcome.

 

Thanks,

 

Alex

 

 

 

 

 

 






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[Asterisk-biz] Audiocodes reseller in Canada ?

[Dave Hawkes]

Hi,

Does anyone know of an Audiocodes reseller in Canada? I'm looking for a mediant 
1000 and am located in Ottawa, ON.

Thanks
Dave Hawkes



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Re: [Asterisk-biz] Phone for Attendant Console (Recepcionist / Operator) - Polycom and other alternatives

[Kevin P. Fleming]

[EMAIL PROTECTED] wrote:

I read at 
http://www.polycom.com/products_services/1,1443,pw-34-182-12104,FF.html 
that it only allows 14 keys... What I am missing ?


You are correct, that's my error :-)
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Re: [Asterisk-biz] Mediatrix config software

[Cory Andrews]

Robert - I am working on getting you the firmware and the management 
tool, don't have the 2102 SIP Image in our archive I am waiting for a 
callback from the vendor.


Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory



Anders Svensson wrote:

If you have bought it new the config software should follow the package.
It's a version for a limited number of units

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: den 28 september 2005 20:57
To: asterisk-biz@lists.digium.com
Subject: [Asterisk-biz] Mediatrix config software


Just purchased a Mediatrix 2102 and am looking for a copy 
of the configuration software. My understanding that it is 
an SNMP based package and does not come with the unit. 
Running SIP protocol.


Thanks,
Robert Webb
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Re: [Asterisk-biz] Phone for Attendant Console (Recepcionist / Operator) - Polycom and other alternatives

[asterisk]

Kevin

I read at http://www.polycom.com/products_services/1,1443,pw-34-182-12104,FF.html
 that it only allows 14 keys... What I am missing ?







"Kevin P. Fleming"
<[EMAIL PROTECTED]> 
Enviado por: [EMAIL PROTECTED]
28/09/2005 19:19



Por favor, responda a
Commercial and Business-Oriented Asterisk Discussion      
 





Para
Commercial and Business-Oriented Asterisk
Discussion <[EMAIL PROTECTED]>


cc



Asunto
Re: [Asterisk-biz] Phone for Attendant
Console (Recepcionist /        Operator) -
Polycom and other alternatives








[EMAIL PROTECTED] wrote:

> It seems a very good option, but unfortunatelly, each extension module
has 
> only 14 keys.. Too few... I would have to add a lot to have the same

> capacity that I would have with a SNOM...So any other suggestions
will 
> be welcome.

That is incorrect; each expansion module provides 20 keys.
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RE: [Asterisk-biz] Mediatrix config software

[Anders Svensson]

If you have bought it new the config software should follow the package.
It's a version for a limited number of units

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: den 28 september 2005 20:57
To: asterisk-biz@lists.digium.com
Subject: [Asterisk-biz] Mediatrix config software


Just purchased a Mediatrix 2102 and am looking for a copy 
of the configuration software. My understanding that it is 
an SNMP based package and does not come with the unit. 
Running SIP protocol.

Thanks,
Robert Webb
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RE: [Asterisk-biz] usa termination

[jay kordic]

I have a .0084 domestic US termination.This is based on a 80/20 termination
into RBOC/NON RBOC areas and requires a $ 1000 monthly commit.

Jay Kordic
The Horizon Group
Wholesale/Retail Voice,Data and IP services
951-780-3131
951-850-0887 (cell)
516-687-5047( fax)
Please ask me about Caller ID compensation

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson
Sent: Tuesday, September 27, 2005 11:20 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: RE: [Asterisk-biz] usa termination

0.008 to 0.009 $

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of wassim darwish
Sent: den 28 september 2005 00:24
To: asterisk-biz@lists.digium.com
Subject: [Asterisk-biz] usa termination

what are the target wholesale rates to usa?

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RE: [Asterisk-biz] Looking for a configuration software

[Kanuri, Seshu \(Company IT\)]

You 
and Jeremy are missing the point, which Jason has clarified.
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel 
BatistaSent: Wednesday, September 28, 2005 2:14 PMTo: 
Commercial and Business-Oriented Asterisk DiscussionSubject: Re: 
[Asterisk-biz] Looking for a configuration software

Amp has changed allot
 
But you can make your own changes and setup in a few files that don't get 
change. Only the voicemail.conf, and the files_additional.conf files get 
changed. There are extensions_custom.conf which you can edit and add your 
settings too. Also you can edit the Sip.conf main area and unless you upgrade 
the system this will not get changed. 
On 9/28/05, Kanuri, Seshu 
(Company IT) <[EMAIL PROTECTED]> 
wrote: 
May 
  be the AMP forum is the right place to post this but, briefly hereare my 
  issues:1)Once AMP is in place, we cannot make changes to the config 
  files using any editor as they are overwritten. This is a major issue as 
  some timesit is esssential to by pass what AMP configures. What is the 
  remedy forthis? Do you have plans to sync-up config files to the DB 
  also?2)AMP does not support Polycom phones now. Even if you add 
  theadditional Polycom related paremeters in the GUI, it does not 
  storethem. For ex:progressinband=no ; 
  Polycom Phones do not support Progressive 
  inband.; 
  We should not use 'progressinband=never'This statement is not saved in 
  version 1.7 or so as I remember. I don'tknow if this has been fixed 
  nowSeshu Kanuri



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RE: [Asterisk-biz] Looking For Asterisk Consultants

[Peter Ericksen]

Garrett's email specified field work, I assume that means actually going to
customer locations (in the field) to install, connect, wire, provision, etc
for VoIP installations. 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara
Sent: Wednesday, September 28, 2005 1:45 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Looking For Asterisk Consultants

Peter Ericksen wrote:

>Where (geographically) will the work be performed?
>  
>


Does geography even matter?95% of the Asterisk Consulting I do is 
via SSH and a Screen session.


Jeremy McNamara
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RE: [Asterisk-biz] Looking for a configuration software

[Kanuri, Seshu \(Company IT\)]

> AMP has customization points

I missed to notice that. Thanks for the clarification.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Wednesday, September 28, 2005 1:59 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Looking for a configuration software

Kanuri, Seshu (Company IT) wrote:
> May be the AMP forum is the right place to post this but, briefly here

> are my issues:

Please do post to the amportal-users mailing list and/or Help forum.
> 
> 1)Once AMP is in place, we cannot make changes to the config files 
> using any editor as they are overwritten. This is a major issue as 
> some times it is esssential to by pass what AMP configures. What is 
> the remedy for this? Do you have plans to sync-up config files to the
DB also?

AMP has customization points.
> 
> 2)AMP does not support Polycom phones now. Even if you add the 
> additional Polycom related paremeters in the GUI, it does not store 
> them. For ex:
> 
>  progressinband=no ; Polycom Phones do not support Progressive 
> inband.
>; We should not use 'progressinband=never'

Many consumers of AMP use Polycom phones - to say that AMP does not
support Polycom phones is patently false. Just about anything that is
not exposed through the interface can be added through a customization
point.

Regards,

Jason


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[Asterisk-biz] Mediatrix config software

[Robert Webb]

Just purchased a Mediatrix 2102 and am looking for a copy 
of the configuration software. My understanding that it is 
an SNMP based package and does not come with the unit. 
Running SIP protocol.


Thanks,
Robert Webb
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[Asterisk-biz] USA/Canada DIDs

[router]

Hi all, 
I need one DID for each area code below:

USA:
408
718
425
773
650
510
212
206
586

Canada:
416
905
519
514
450
604

I need incoming only, unlimited, multiple channels per DID, SIP or IAX.
More cities to follow.

Please contact me off list with quotes.

Thank you.


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RE: [Asterisk-biz] Looking for a configuration software

[Colin Anderson]

>1)Once AMP is in place, we cannot make changes to the config files using
>any editor as they are overwritten. This is a major issue as some times
>it is esssential to by pass what AMP configures. What is the remedy for
>this? Do you have plans to sync-up config files to the DB also?

I create the extension in AMP which writes the sip.conf correctly and
creates the voicemail box, then in extensions.conf:

[from-pstn]

include => mycustomcontext

and in extensions_custom.conf:

[mycustomcontext]

exten => myextension,s,1,DoStuff()

As long as the include is at the top of the from-pstn context, it is parsed
before hitting the AMP stuff, thereby bypassing any AMP logic. AMP is good
about not fooling around with includes in extensions.conf and leaves them
alone. extensions_custom.conf AMP leaves alone 100%.

Reason I do it this way is I don't trust my fingers to not make a typo in
sip.conf and voicemail.conf. 

Only drawback to this is sometimes you and AMP have to dance around each
other for precisely the reason you mention and eventually you have spaghetti
conf files from hell. I don't know how this could be done otherwise though
with a flat file. 

hth
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Re: [Asterisk-biz] Looking for a configuration software

[Ariel Batista]

Amp has changed allot
 
But you can make your own changes and setup in a few files that don't get change. Only the voicemail.conf, and the files_additional.conf files get changed. There are extensions_custom.conf which you can edit and add your settings too. Also you can edit the 
Sip.conf main area and unless you upgrade the system this will not get changed. 
On 9/28/05, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote:
May be the AMP forum is the right place to post this but, briefly hereare my issues:1)Once AMP is in place, we cannot make changes to the config files using
any editor as they are overwritten. This is a major issue as some timesit is esssential to by pass what AMP configures. What is the remedy forthis? Do you have plans to sync-up config files to the DB also?
2)AMP does not support Polycom phones now. Even if you add theadditional Polycom related paremeters in the GUI, it does not storethem. For ex:progressinband=no ; Polycom Phones do not support Progressive
inband.; We should not use 'progressinband=never'This statement is not saved in version 1.7 or so as I remember. I don'tknow if this has been fixed nowSeshu Kanuri
-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]
] On Behalf Of Jason BeckerSent: Wednesday, September 28, 2005 12:10 PMTo: Commercial and Business-Oriented Asterisk DiscussionSubject: Re: [Asterisk-biz] Looking for a configuration softwareColin Anderson wrote:
> Demo looks very good. I like how you de-aggregate all of the various> functions i.e. you can add *just* a voicemail or *just* a phone, with> AMP it's kind of all or nothing. The multi-tenant is great.
Just wanted to point out that with the release of 1.10.009 you do havemore control over this. You can create just a User (and optionallyVoicemailbox) or just a Device. And the Device can be any Asterisk
supported technology. As an example I can create a Zap Device(Zap/[channel]/XXX) and associate it with my User (Extension) sothat when I am out of the office when someone dials my extension it willring my cel. When I'm back in the office I can turn this off so that my
desk phone rings when my extension is dialed. The Device/User separationin 009 lays the foundation for features like find-me/follow-me, etc.Regards,--Jason BeckerDirector & CEOCoalescent Systems Inc.
Enabling Open Source Telephony403.244.8089www.coalescentsystems.ca___Asterisk-Biz mailing list
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-- Ariel BatistaKasi InternationalOffice: 305-574-6721  FX: 305-574-0212Home: 305-574-1495 Cell: 786-376-1896 
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Re: [Asterisk-biz] Looking for a configuration software

[Jeremy McNamara]

Kanuri, Seshu (Company IT) wrote:


May be the AMP forum is the right place to post this but, briefly here
are my issues:

1)Once AMP is in place, we cannot make changes to the config files using
any editor as they are overwritten. This is a major issue as some times
it is esssential to by pass what AMP configures. What is the remedy for
this? Do you have plans to sync-up config files to the DB also?

 



If the lame developers of AMP would find a clue, this would not be a 
problem.


There is a wonderful feature called #include - read about it.


Jeremy McNamara
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Re: [Asterisk-biz] Looking for a configuration software

[Jason Becker]

Kanuri, Seshu (Company IT) wrote:

May be the AMP forum is the right place to post this but, briefly here
are my issues:


Please do post to the amportal-users mailing list and/or Help forum.


1)Once AMP is in place, we cannot make changes to the config files using
any editor as they are overwritten. This is a major issue as some times
it is esssential to by pass what AMP configures. What is the remedy for
this? Do you have plans to sync-up config files to the DB also?


AMP has customization points.


2)AMP does not support Polycom phones now. Even if you add the
additional Polycom related paremeters in the GUI, it does not store
them. For ex:

 progressinband=no ; Polycom Phones do not support Progressive
inband. 
			 ; We should not use 'progressinband=never'


Many consumers of AMP use Polycom phones - to say that AMP does not 
support Polycom phones is patently false. Just about anything that is 
not exposed through the interface can be added through a customization 
point.


Regards,

Jason



This statement is not saved in version 1.7 or so as I remember. I don't
know if this has been fixed now

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Wednesday, September 28, 2005 12:10 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Looking for a configuration software

Colin Anderson wrote:

Demo looks very good. I like how you de-aggregate all of the various 
functions i.e. you can add *just* a voicemail or *just* a phone, with 
AMP it's kind of all or nothing. The multi-tenant is great.



Just wanted to point out that with the release of 1.10.009 you do have
more control over this. You can create just a User (and optionally
Voicemailbox) or just a Device. And the Device can be any Asterisk
supported technology. As an example I can create a Zap Device
(Zap/[channel]/XXX) and associate it with my User (Extension) so
that when I am out of the office when someone dials my extension it will
ring my cel. When I'm back in the office I can turn this off so that my
desk phone rings when my extension is dialed. The Device/User separation
in 009 lays the foundation for features like find-me/follow-me, etc.

Regards,

--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-biz] Looking For Asterisk Consultants

[Jeremy McNamara]

Peter Ericksen wrote:


Where (geographically) will the work be performed?
 




Does geography even matter?95% of the Asterisk Consulting I do is 
via SSH and a Screen session.



Jeremy McNamara
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Re: [Asterisk-biz] Looking For Asterisk Consultants

[Michael Welter]

Drat that pesky reply button.



Garrett Smith wrote:

All:

 

I am working on a project that needs Asterisk consultants to do some 
field work. I am looking for 3-5 consultants that are willing to do some 
on-site work. The extent and scope is yet to be exactly pinned down, but 
you would be paid a lump fee for work performed. You would also be 
working as an agent and or employee of VoIPSupply.com, representing our 
company, not your own. If anyone is interested in this, please send me 
your resume, as well general wage requirements (hourly, daily, weekly) 
and I will send more specific information later this week.


 


Thank you all!

 

 


Garrett Smith

[EMAIL PROTECTED] 

 


Internet Telephony Expo Booth # 623!

 


VoIPSupply.com

 -- a division of B2 Technologies, LLC

 


(716) 250-3408 Direct

(716) 630-1548 Fax

(716) 903-9495 Cell

 


AOL IM: B2sales

 

VoIPSupply.com is North America's largest VAR of VoIP Hardware. We offer 
the Internet's largest online product catalog featuring products from 
Sipura, Linksys, Cisco, Polycom, Digium, Grandstream, Snom, Mediatrix, 
Pingtel, Centrepoint Technologies, Carrier Access, Clipcomm, and many more!


 

 





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Re: [Asterisk-biz] Looking For Asterisk Consultants

[Michael Welter]

Hello Garrett,

I am available for short-term projects.  My rate is $110/hr plus 
expenses, and I can propose a fixed price if you'll send me the project 
details.


I have been involved with VoIP and Asterisk since early 2003.  I have 10 
in-house Asterisk installations in the Denver area.  I also have a 
hosted Asterisk system which serves several clients.  This Friday I'm 
installing an Asterisk system using (your) 7960 telephone sets.


My installations include analog telephone sets with Adtran 750 channel 
banks.  My most recent installations utilize only SIP phones.  I have 
experience with T1, both PRI and CAS.


My background is software development.  For one of my Denver clients 
I've developed an extensive IVR-based data collection system for 
visiting nurses and homecare workers.  I also developed their web site 
for non-IVR data collection and reporting.


From 1978 until 2003 I developed software for the brokerage and banking 
industry.  I'm fluent in Java, C, PHP, Perl, HTML, and XML.  My company, 
Introspect Consulting, was incorporated in 1983, and we're going to 
change the name to TelecomMatters.net .


From 1969 until 1978 I developed software for the telephone industry.

During the late 60s I was an officer in the US Army.

I'll be at Astricon--maybe we can meet for a chat.

Thank you for your attention.
Michael Welter
303-718-2804





Garrett Smith wrote:

All:

 

I am working on a project that needs Asterisk consultants to do some 
field work. I am looking for 3-5 consultants that are willing to do some 
on-site work. The extent and scope is yet to be exactly pinned down, but 
you would be paid a lump fee for work performed. You would also be 
working as an agent and or employee of VoIPSupply.com, representing our 
company, not your own. If anyone is interested in this, please send me 
your resume, as well general wage requirements (hourly, daily, weekly) 
and I will send more specific information later this week.


 


Thank you all!

 

 


Garrett Smith

[EMAIL PROTECTED] 

 


Internet Telephony Expo Booth # 623!

 


VoIPSupply.com

 -- a division of B2 Technologies, LLC

 


(716) 250-3408 Direct

(716) 630-1548 Fax

(716) 903-9495 Cell

 


AOL IM: B2sales

 

VoIPSupply.com is North America's largest VAR of VoIP Hardware. We offer 
the Internet's largest online product catalog featuring products from 
Sipura, Linksys, Cisco, Polycom, Digium, Grandstream, Snom, Mediatrix, 
Pingtel, Centrepoint Technologies, Carrier Access, Clipcomm, and many more!


 

 





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RE: [Asterisk-biz] Attendant Phone

[Colin Anderson]

There is a SIP stack for a 5220 and 5235 which the sidecar attaches to but
it is unclear as to whether it will work in SIP mode. We have one that is
running MiNet and once I get rid of my 3300 I will experiment.
Unfortunately, MiNet is here at my install for the remainder of 2005.

-Original Message-
From: Peter Ericksen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 28, 2005 10:31 AM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz] Attendant Phone


I was looking for an attendant phone myself a couple weeks ago, and still
haven't found anything suitable.  I need something that can grow to 200
DSS/BLF keys.
 
Looking today I did find that Mitel makes an attendant console that will go
up to 96 keys, does anyone know if it will work with Asterisk?
 
http://www.mitel.com/DocController?documentId=9806&c=9512&sc=9518

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[Asterisk-biz] VoIP engineer needed

[Robert Bedell]

I hope this is the correct place to
post this.  I’m not certain if it’s kosher to post to –users
as well.

 

 

Company: ZivVa

Location: Atlanta, GA

Position: Software Engineer

Pay: depending on qualifications

 

General Description:  ZivVa is
a VoIP startup who is looking for a self motivated Software Engineer who will
thrive in a startup environment.  The programming duties will vary across
database, c/c++, linux, and web development.  Our philosophy is for the development
team to work closely with the customer so as to get first hand experience in
the problems/solutions.  The position will include programming, system
support and troubleshooting customer issues.  You will be working closely with
Asterisk in a hosted environment.  This is strictly VoIP work, but familiarity
with traditional telephony is a plus.

Qualifications:
- Detailed knowledge of the Linux operating
system 

- Programming skills in C,PHP, Perl,
MySQL, and PostGreSQL
- Knowledge of basic computer hardware
installation/troubleshooting, DHCP, DNS, and basic TCP/IP
- VoIP and Asterisk experience required

- Comfortable interacting with
customers by phone, e-mail and face to face

 

Please contact me off list to inquire about this position,
or send mail to [EMAIL PROTECTED].

 

Robert Bedell

Chief Technical Officer

ZivVa, LLC

 

 

 






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Re: [Asterisk-biz] Attendant Phone

[Kevin P. Fleming]

Peter Ericksen wrote:

I was looking for an attendant phone myself a couple weeks ago, and still
haven't found anything suitable.  I need something that can grow to 200
DSS/BLF keys.


This will be completely impractical for Asterisk use; the only method 
SIP devices have of keeping DSS/BLF keys updated is via SIP 
SUBSCRIBE/NOTIFY, and the overhead of handling 200 of them on a single 
phone will be enormous.


For that scale, you need to consider PC-based solutions, not phones.
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Re: [Asterisk-biz] Phone for Attendant Console (Recepcionist / Operator) - Polycom and other alternatives

[Kevin P. Fleming]

[EMAIL PROTECTED] wrote:

It seems a very good option, but unfortunatelly, each extension module has 
only 14 keys.. Too few... I would have to add a lot to have the same 
capacity that I would have with a SNOM...So any other suggestions will 
be welcome.


That is incorrect; each expansion module provides 20 keys.
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RE: [Asterisk-biz] Attendant Phone

[Derrick Moennick]

How about the snom 220?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Ericksen
Sent: Wednesday, September 28, 2005 9:31 AM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz] Attendant Phone

I was looking for an attendant phone myself a couple weeks ago, and
still
haven't found anything suitable.  I need something that can grow to 200
DSS/BLF keys.
 
Looking today I did find that Mitel makes an attendant console that will
go
up to 96 keys, does anyone know if it will work with Asterisk?
 
http://www.mitel.com/DocController?documentId=9806&c=9512&sc=9518

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RE: [Asterisk-biz] Looking for a configuration software

[Kanuri, Seshu \(Company IT\)]

May be the AMP forum is the right place to post this but, briefly here
are my issues:

1)Once AMP is in place, we cannot make changes to the config files using
any editor as they are overwritten. This is a major issue as some times
it is esssential to by pass what AMP configures. What is the remedy for
this? Do you have plans to sync-up config files to the DB also?

2)AMP does not support Polycom phones now. Even if you add the
additional Polycom related paremeters in the GUI, it does not store
them. For ex:

 progressinband=no ; Polycom Phones do not support Progressive
inband. 
 ; We should not use 'progressinband=never'

This statement is not saved in version 1.7 or so as I remember. I don't
know if this has been fixed now

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Wednesday, September 28, 2005 12:10 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Looking for a configuration software

Colin Anderson wrote:
> Demo looks very good. I like how you de-aggregate all of the various 
> functions i.e. you can add *just* a voicemail or *just* a phone, with 
> AMP it's kind of all or nothing. The multi-tenant is great.

Just wanted to point out that with the release of 1.10.009 you do have
more control over this. You can create just a User (and optionally
Voicemailbox) or just a Device. And the Device can be any Asterisk
supported technology. As an example I can create a Zap Device
(Zap/[channel]/XXX) and associate it with my User (Extension) so
that when I am out of the office when someone dials my extension it will
ring my cel. When I'm back in the office I can turn this off so that my
desk phone rings when my extension is dialed. The Device/User separation
in 009 lays the foundation for features like find-me/follow-me, etc.

Regards,

--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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RE: [Asterisk-biz] Looking For Asterisk Consultants

[Peter Ericksen]

Where (geographically) will the work be performed?
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garrett Smith
Sent: Wednesday, September 28, 2005 12:02 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz] Looking For Asterisk Consultants



All:

 

I am working on a project that needs Asterisk consultants to do some field
work. I am looking for 3-5 consultants that are willing to do some on-site
work. The extent and scope is yet to be exactly pinned down, but you would
be paid a lump fee for work performed. You would also be working as an agent
and or employee of VoIPSupply.com, representing our company, not your own.
If anyone is interested in this, please send me your resume, as well general
wage requirements (hourly, daily, weekly) and I will send more specific
information later this week.

 

Thank you all!

 

 

Garrett Smith

[EMAIL PROTECTED]  

 

Internet Telephony Expo Booth # 623!

 

VoIPSupply.com

 -- a division of B2 Technologies, LLC

 

(716) 250-3408 Direct

(716) 630-1548 Fax

(716) 903-9495 Cell

 

AOL IM: B2sales

 

VoIPSupply.com is North America's largest VAR of VoIP Hardware. We offer the
Internet's largest online product catalog featuring products from Sipura,
Linksys, Cisco, Polycom, Digium, Grandstream, Snom, Mediatrix, Pingtel,
Centrepoint Technologies, Carrier Access, Clipcomm, and many more!

 

 


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Checked by AVG Anti-Virus.
Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/2005



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[Asterisk-biz] Attendant Phone

[Peter Ericksen]

I was looking for an attendant phone myself a couple weeks ago, and still
haven't found anything suitable.  I need something that can grow to 200
DSS/BLF keys.
 
Looking today I did find that Mitel makes an attendant console that will go
up to 96 keys, does anyone know if it will work with Asterisk?
 
http://www.mitel.com/DocController?documentId=9806&c=9512&sc=9518

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[Asterisk-biz] Phone for Attendant Console (Recepcionist / Operator) - Polycom and other alternatives

[asterisk]

Hi,

Looking for an Phone for Attendant Console
(Recepcionist / Operator)   I only was able to found this product
from a previos treat:

http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,4885,00.pdf

It seems a very good option, but unfortunatelly,
each extension module has only 14 keys.. Too few... I would have to add
a lot to have the same capacity that I would have with a SNOM...So
any other suggestions will be welcome.

However, I am desesperate to find a
fast alternative, and if not find any one soon, I will try to adquire this
model the fastest way. Before doing that  my questions are:

a).- Has anyone tested this model with
Asterisk ?  I am afraid to buy and expensive model that will have
to be shipped the fastest way from USA and not having any guarantee.

b).- Seems that this model is very,
very new...I observe some post that told this model is not still shipped,
but  I find it on www.voipsupply.com, and they seems to have it on
stock (is on the shopping web center).  am I missing anything here
?

c).- Any other altenartive ? My wholeshale
on Spain suggested me to use the avaya model 5620, and then, I checked
his web page and observe it was NOT compatible with SIP (it only worked
on their own PBX).  So, people here seems does not to known real alternatives.
Even worse, it seems there is to few of them.
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Re: [Asterisk-biz] Looking for a configuration software

[Jason Becker]

Colin Anderson wrote:

Demo looks very good. I like how you de-aggregate all of the various
functions i.e. you can add *just* a voicemail or *just* a phone, with AMP
it's kind of all or nothing. The multi-tenant is great. 


Just wanted to point out that with the release of 1.10.009 you do have 
more control over this. You can create just a User (and optionally 
Voicemailbox) or just a Device. And the Device can be any Asterisk 
supported technology. As an example I can create a Zap Device 
(Zap/[channel]/XXX) and associate it with my User (Extension) so 
that when I am out of the office when someone dials my extension it will 
ring my cel. When I'm back in the office I can turn this off so that my 
desk phone rings when my extension is dialed. The Device/User separation 
in 009 lays the foundation for features like find-me/follow-me, etc.


Regards,

--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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[Asterisk-biz] Looking For Asterisk Consultants

[Garrett Smith]

All:

 

I am working on a project that needs Asterisk
consultants to do some field work. I am looking for 3-5 consultants that are
willing to do some on-site work. The extent and scope is yet to be exactly pinned
down, but you would be paid a lump fee for work performed. You would also be
working as an agent and or employee of VoIPSupply.com, representing our company,
not your own. If anyone is interested in this, please send me your resume, as
well general wage requirements (hourly, daily, weekly) and I will send more
specific information later this week.

 

Thank you all!

 

 

Garrett Smith

[EMAIL PROTECTED]

 

Internet Telephony Expo Booth # 623!

 

VoIPSupply.com

 -- a division of B2 Technologies, LLC

 

(716) 250-3408 Direct

(716) 630-1548 Fax

(716) 903-9495 Cell

 

AOL IM: B2sales

 

VoIPSupply.com is North America's
largest VAR of VoIP Hardware. We offer the Internet's largest online product
catalog featuring products from Sipura, Linksys, Cisco, Polycom, Digium,
Grandstream, Snom, Mediatrix, Pingtel, Centrepoint Technologies, Carrier
Access, Clipcomm, and many more!

 

 






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RE: [Asterisk-biz] Serious Problem with SNOM Attendant phones -NeedHelp

[canuck15]

I am curious, are these general problems with the SNOM 360 or only when you
put the sidecar on or ?  As others have said, this is currently the only
option for a receptionist type scenario with many extensions that need
watching.  I thought this phone worked relatively well for that based on
what I have read at various sites. 

> -Original Message-
> From: Jerry Jones [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, September 28, 2005 6:51 AM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [Asterisk-biz] Serious Problem with SNOM 
> Attendant phones -NeedHelp
> 
> Polycom is not yet shipping - at least on Monday when I last checked.
> 
> We too had so many problems with Snoms in this situation. we 
> have stopped using, and most customers want them replaced 
> with Polycoms.
> 
> We started using FOP as a display with a dedicated computer 
> and flat screen for operators. In most cases the customer 
> just wants to see who is on the line and it works very well 
> for this. We do not even install with keyboards and mouse so 
> they cannot use for anything other than display. I am hopeful 
> the new Polycom sidecars - which only work with the IP601 - 
> will be succesful.
> 
> On Sep 28, 2005, at 8:08 AM, Dean Collins wrote:
> 
> > Polycom have an attendant phone, not sure how large it can 
> expand to 
> > but I've seen a 3 car photo of one.
> >
> >
> >
> > Cheers,
> >
> > Dean
> >
> >
> >
> >
> >
> >
> >
> > From: [EMAIL PROTECTED] [mailto:asterisk-biz- 
> > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> > Sent: Wednesday, 28 September 2005 8:47 AM
> > To: asterisk-biz@lists.digium.com
> > Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]
> > Subject: [Asterisk-biz] Serious Problem with SNOM Attendant phones
> > - NeedHelp
> > Importance: High
> >
> >
> >
> >
> > Hi,
> >
> > One of my biggest customer, with more than 300 hundred persons  
> > using asterisk, is having bing problems with the SNOM phones used  
> > at the operator position (Attendant Console). We currently have 3  
> > operator attending incoming calls and bypassing them to the 
> rest of  
> > the company.
> >
> > We did not find any alternative for SNOM to use as Attendant  
> > Console (phone for the operator). We need Operators to have a  
> > keypad where the could observe the most important 
> extensions of the  
> > office, and be able to transfer a call very fast (just pressing a  
> > button).
> >
> > Recently, SNOM released a new model. The SNOM-360 phone, that was  
> > ideal for that.  However, it only supports one keypad (around 30  
> > extensions), althought previous models were able to support 
> several  
> > of them. SNOM tolds us they are working on this, but they 
> would not  
> > have a solution until end of the year (at least). We had pay in  
> > advance for the models a lot of time ago, but it seems there is a  
> > lot of problems with SNOM stock, and they did not got units to  
> > server us. Finally, and due a hard preassure, SNOM sent us 
> directly  
> > 3 used phones they got in their lab. We received the password of  
> > only 2 of them.
> >
> > A week after being in production, we started with a lot of 
> problems  
> > with them. Such as phones reboot themselfes each 10 minutes. We  
> > tryed to upload a more recent firmware, but one of them got  
> > broken.  We have NOW in production just on phone, that still  
> > reboots each 10 minutes.  Problem is on the SNOM phone, since with  
> > all the phones we have connected to Asterisk everything goes ok.
> >
> > Customer is unable to attend all theirs calls, and is really  
> > upset.  I need to react very fast.
> >
> > Trying to anticipate to this, one of our developers did a program  
> > for the Operator. That program used the Asterisk Manager API of  
> > asterisk in order to check events, and allow the operator to use a  
> > program to chek extension status and forward the call to any  
> > extension very fast. However, it seems that Asterisk becomes  
> > unstable when has a lot of traffic and your are using one of those  
> > programas, and having that program installed caused a lot of  
> > crashes (we think Asterisk Manager API is not very stable, so  
> > far).  We do not like FOP because it is complicated to use for an  
> > untrained operator, that could easily lost calls, and do incorrect  
> > operations. We need a more powerfull software for this.
> >
> > I saw that 3COM has also attendace phones with a lot of 
> extensions,  
> > but they do not distribute yet that phone on Europe.  I 
> need a very  
> > fast and urgent solution for my operators, and have a phone for  
> > them asap.
> >
> > Could any one, please, suggest me any solution ? Any one could  
> > serve me really fast a Attendance Phone, with a lot of extensions,  
> > for using on Asterisk to deal with this problem? (please, no more  
> > SNOMs, customer does not want further to see them).
> >
> > Since this problem is really urgent I

[Asterisk-biz] Consultant Help

[Tavino, Ralph]

Title: Consultant Help





I'm still interested in hearing from anyone interested in helping with project-specific support (on a paid consultant basis, of course).  We currently have one project in progress in the LA area that we'd like to discuss.

Ralph Tavino
Director, Customer Support
Carrier Access
[EMAIL PROTECTED]
303.218.5579
Customer Support Center:  800.786.9929




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Re: [Asterisk-biz] Looking for a configuration software

[Dustin Wildes]

Thank you!  :-)

Hopefully all of the loose ends will be wrapped up with version 2.7
coming out in a week or so.
We'll also be working on a SER integration as well to have one
interface.

Also - the security manager will also be coming in 2.7  which will
allow users to admin their own account, review calls they made, change
voicemail, listen to voicemail, add call-forward options, etc...  along
with local admin access.  The current version is the 'super-user' or
root-type configuration tool.



Colin Anderson wrote:

  Demo looks very good. I like how you de-aggregate all of the various
functions i.e. you can add *just* a voicemail or *just* a phone, with AMP
it's kind of all or nothing. The multi-tenant is great. Well done. Tie up
your loose ends and you've got a great product there. 

-Original Message-
From: Dustin Wildes [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, September 27, 2005 9:46 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Looking for a configuration software


I'd like to throw our product out there as well - PhoneCALL.
You can find information here:
http://www.vecsector.com/phonecall

Demo Here:
http://www.vecsector.com/phonecall/demo
Username:  demo
Password:  demo

We are in the very last stages of building a reseller & OEM channel, 
among different support options.
It's completely PHP/MySQL-based and built with the SMARTY template 
engine, so specific hardware is required.
We support multi-tenant, and a very advanced scripting engine.

We're also about to release version 2.7 which addresses the hardware 
configuration problems that have been reported to us.
In addition, we will be releasing a commercially support embedded 
product with Asterisk & PhoneCALL combined, among lots of other common 
applications setup in solid state.

Thanks!!


Dustin Wildes
VecSector, LLC

M. MAINGAULT wrote:

  
  
Hi,
i'm looking for a configuration tool for asterisk (with a lot of 
options and features).
I already know PBXware of Bicom Systems and xorcom rapid configuration 
tool.
Xorcom is good but the problem is that there is not a lot of 
configurations options. Price is good.
PBXware is too expensive.
 
I'll be very interested in a software who can generate configurations 
files for asterisk : i would like pay once for the software, but not 
paid for each configurations (except if the price is low...)
 
Best regards,
Mr. MAINGAULT
CEO
ITS
France



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Re: [Asterisk-biz] Serious Problem with SNOM Attendant phones - Need Help

[Patrick]

On Wed, 2005-09-28 at 14:47 +0200, [EMAIL PROTECTED] wrote:
> 
> Hi, 

Hi,

I hope you have called Snom about these issues and worked with them to
get it resolved? If you need a bigger Operator phone then give Polycom a
call or have a look here:
http://www.polycom.com/products_services/1,,pw-34-182-12072,00.html

Regard,
Patrick
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[Asterisk-biz] seeking an Asterisk expert

[Fabian Müller]

Hello,

we have a job offer for readers located in Germany.

We are seeking an Asterisk expert with very good knowledge in standard
VoIP protocols and long experience in developing software in C for
Linux.

You will develop Asterisk applications and install and administrate a
failsafe Asterisk installation with several servers that will be able
to receive a few hundred calls at the same time.

You should know "something" about SER as well.

If you know that you can manage this I would be happy to hear from you
and receive an email at [EMAIL PROTECTED] with information
about your experiences in the relevant fields.

For further questions contact me off list please.

Greetings,

Fabian Müller
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Re: [Asterisk-biz] PAP2-NA New or Used

[Bruce Ferrell]

Website says you have to order 5 or they cancel your order :(

Danny Froberg wrote:
Check with Cory & Co at voipsupply.com, I think they stacked up a bunch 
of these.


/Danny

Michael Blood wrote:

Does anyone have any PAP2-NA available for sale.  I konw they are on 
back order everywhere but I thought I would send out a quick question.
 
You can reply to me directly at

[EMAIL PROTECTED] 
 
Thanks
 
Michael Blood





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RE: [Asterisk-biz] Serious Problem with SNOM Attendant phones - N eedHelp

[Colin Anderson]

This won't address OP's problem, but in my install I had to grapple with
weakness in the operator position (yes we use SNOM's not afraid to admit it,
really like them) and to deal with that I had to change our corporate
culture somewhat. 

The solution was to simply assign every user a DID and inform them that
"this is your personal phone number / fax number, it will follow you
wherever you are for the term of your employment here" and let them do the
rest. Once they got the word out to the dozen or so key people that each of
them talk to every day, call volume dropped to the operator position by 90%.
In this case everyone wins:

-Caller feels "important" because they have a personal phone number to call
and they don't have to go through reception

-Through dialplan magic, call will chase the user i.e. if they are offsite
call is rerouted to their cell if they don't pick up

-Operator was *so* happy that she didn't mind having to "Xfer" --> 4 digit
exten for the couple dozen calls she got during the day, this to her was way
better than picking up the phone 300-400 times a day, as well it freed her
up to take a pee break and the like. 

-Our PBX has the appearance of being more robust from this functionality
even though fundamentally I am working around a limitation. 

Gawd, I love Asterisk. 

-Original Message-
From: Jerry Jones [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 28, 2005 7:51 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Serious Problem with SNOM Attendant phones -
NeedHelp


Polycom is not yet shipping - at least on Monday when I last checked.

We too had so many problems with Snoms in this situation. we have  
stopped using, and most customers want them replaced with Polycoms.

We started using FOP as a display with a dedicated computer and flat  
screen for operators. In most cases the customer just wants to see  
who is on the line and it works very well for this. We do not even  
install with keyboards and mouse so they cannot use for anything  
other than display. I am hopeful the new Polycom sidecars - which  
only work with the IP601 - will be succesful.

On Sep 28, 2005, at 8:08 AM, Dean Collins wrote:

> Polycom have an attendant phone, not sure how large it can expand  
> to but I’ve seen a 3 car photo of one.
>
>
>
> Cheers,
>
> Dean
>
>
>
>
>
>
>
> From: [EMAIL PROTECTED] [mailto:asterisk-biz- 
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Wednesday, 28 September 2005 8:47 AM
> To: asterisk-biz@lists.digium.com
> Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]
> Subject: [Asterisk-biz] Serious Problem with SNOM Attendant phones  
> - NeedHelp
> Importance: High
>
>
>
>
> Hi,
>
> One of my biggest customer, with more than 300 hundred persons  
> using asterisk, is having bing problems with the SNOM phones used  
> at the operator position (Attendant Console). We currently have 3  
> operator attending incoming calls and bypassing them to the rest of  
> the company.
>
> We did not find any alternative for SNOM to use as Attendant  
> Console (phone for the operator). We need Operators to have a  
> keypad where the could observe the most important extensions of the  
> office, and be able to transfer a call very fast (just pressing a  
> button).
>
> Recently, SNOM released a new model. The SNOM-360 phone, that was  
> ideal for that.  However, it only supports one keypad (around 30  
> extensions), althought previous models were able to support several  
> of them. SNOM tolds us they are working on this, but they would not  
> have a solution until end of the year (at least). We had pay in  
> advance for the models a lot of time ago, but it seems there is a  
> lot of problems with SNOM stock, and they did not got units to  
> server us. Finally, and due a hard preassure, SNOM sent us directly  
> 3 used phones they got in their lab. We received the password of  
> only 2 of them.
>
> A week after being in production, we started with a lot of problems  
> with them. Such as phones reboot themselfes each 10 minutes. We  
> tryed to upload a more recent firmware, but one of them got  
> broken.  We have NOW in production just on phone, that still  
> reboots each 10 minutes.  Problem is on the SNOM phone, since with  
> all the phones we have connected to Asterisk everything goes ok.
>
> Customer is unable to attend all theirs calls, and is really  
> upset.  I need to react very fast.
>
> Trying to anticipate to this, one of our developers did a program  
> for the Operator. That program used the Asterisk Manager API of  
> asterisk in order to check events, and allow the operator to use a  
> program to chek extension status and forward the call to any  
> extension very fast. However, it seems that Asterisk becomes  
> unstable when has a lot of traffic and your are using one of those  
> programas, and having that program installed caused a lot of  
> crashes (we think Asterisk Manager API is not v

RE: [Asterisk-biz] Looking for a configuration software

[Colin Anderson]

Demo looks very good. I like how you de-aggregate all of the various
functions i.e. you can add *just* a voicemail or *just* a phone, with AMP
it's kind of all or nothing. The multi-tenant is great. Well done. Tie up
your loose ends and you've got a great product there. 

-Original Message-
From: Dustin Wildes [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 27, 2005 9:46 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-biz] Looking for a configuration software


I'd like to throw our product out there as well - PhoneCALL.
You can find information here:
http://www.vecsector.com/phonecall

Demo Here:
http://www.vecsector.com/phonecall/demo
Username:  demo
Password:  demo

We are in the very last stages of building a reseller & OEM channel, 
among different support options.
It's completely PHP/MySQL-based and built with the SMARTY template 
engine, so specific hardware is required.
We support multi-tenant, and a very advanced scripting engine.

We're also about to release version 2.7 which addresses the hardware 
configuration problems that have been reported to us.
In addition, we will be releasing a commercially support embedded 
product with Asterisk & PhoneCALL combined, among lots of other common 
applications setup in solid state.

Thanks!!


Dustin Wildes
VecSector, LLC

M. MAINGAULT wrote:

> Hi,
> i'm looking for a configuration tool for asterisk (with a lot of 
> options and features).
> I already know PBXware of Bicom Systems and xorcom rapid configuration 
> tool.
> Xorcom is good but the problem is that there is not a lot of 
> configurations options. Price is good.
> PBXware is too expensive.
>  
> I'll be very interested in a software who can generate configurations 
> files for asterisk : i would like pay once for the software, but not 
> paid for each configurations (except if the price is low...)
>  
> Best regards,
> Mr. MAINGAULT
> CEO
> ITS
> France
>
>
>
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>  
>

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Re: [Asterisk-biz] Serious Problem with SNOM Attendant phones - NeedHelp

[Jerry Jones]

Polycom is not yet shipping - at least on Monday when I last checked.

We too had so many problems with Snoms in this situation. we have  
stopped using, and most customers want them replaced with Polycoms.


We started using FOP as a display with a dedicated computer and flat  
screen for operators. In most cases the customer just wants to see  
who is on the line and it works very well for this. We do not even  
install with keyboards and mouse so they cannot use for anything  
other than display. I am hopeful the new Polycom sidecars - which  
only work with the IP601 - will be succesful.


On Sep 28, 2005, at 8:08 AM, Dean Collins wrote:

Polycom have an attendant phone, not sure how large it can expand  
to but I’ve seen a 3 car photo of one.




Cheers,

Dean







From: [EMAIL PROTECTED] [mailto:asterisk-biz- 
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]

Sent: Wednesday, 28 September 2005 8:47 AM
To: asterisk-biz@lists.digium.com
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-biz] Serious Problem with SNOM Attendant phones  
- NeedHelp

Importance: High




Hi,

One of my biggest customer, with more than 300 hundred persons  
using asterisk, is having bing problems with the SNOM phones used  
at the operator position (Attendant Console). We currently have 3  
operator attending incoming calls and bypassing them to the rest of  
the company.


We did not find any alternative for SNOM to use as Attendant  
Console (phone for the operator). We need Operators to have a  
keypad where the could observe the most important extensions of the  
office, and be able to transfer a call very fast (just pressing a  
button).


Recently, SNOM released a new model. The SNOM-360 phone, that was  
ideal for that.  However, it only supports one keypad (around 30  
extensions), althought previous models were able to support several  
of them. SNOM tolds us they are working on this, but they would not  
have a solution until end of the year (at least). We had pay in  
advance for the models a lot of time ago, but it seems there is a  
lot of problems with SNOM stock, and they did not got units to  
server us. Finally, and due a hard preassure, SNOM sent us directly  
3 used phones they got in their lab. We received the password of  
only 2 of them.


A week after being in production, we started with a lot of problems  
with them. Such as phones reboot themselfes each 10 minutes. We  
tryed to upload a more recent firmware, but one of them got  
broken.  We have NOW in production just on phone, that still  
reboots each 10 minutes.  Problem is on the SNOM phone, since with  
all the phones we have connected to Asterisk everything goes ok.


Customer is unable to attend all theirs calls, and is really  
upset.  I need to react very fast.


Trying to anticipate to this, one of our developers did a program  
for the Operator. That program used the Asterisk Manager API of  
asterisk in order to check events, and allow the operator to use a  
program to chek extension status and forward the call to any  
extension very fast. However, it seems that Asterisk becomes  
unstable when has a lot of traffic and your are using one of those  
programas, and having that program installed caused a lot of  
crashes (we think Asterisk Manager API is not very stable, so  
far).  We do not like FOP because it is complicated to use for an  
untrained operator, that could easily lost calls, and do incorrect  
operations. We need a more powerfull software for this.


I saw that 3COM has also attendace phones with a lot of extensions,  
but they do not distribute yet that phone on Europe.  I need a very  
fast and urgent solution for my operators, and have a phone for  
them asap.


Could any one, please, suggest me any solution ? Any one could  
serve me really fast a Attendance Phone, with a lot of extensions,  
for using on Asterisk to deal with this problem? (please, no more  
SNOMs, customer does not want further to see them).


Since this problem is really urgent I will appreciate if any one  
having any proposal could contact directly with me at andres at  
europesip.com. Any suggestions of specific phones with keypads (to  
be used as Attendance Phones) or programs we could use for the  
operator, will help a lot.


kind Regards,

Andres


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RE: [Asterisk-biz] Serious Problem with SNOM Attendant phones - NeedHelp

[Dean Collins]

Polycom have an attendant phone, not sure
how large it can expand to but I’ve seen a 3 car photo of one.

 

Cheers,

Dean

 

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, 28 September 2005
8:47 AM
To: asterisk-biz@lists.digium.com
Cc: [EMAIL PROTECTED];
[EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: [Asterisk-biz] Serious
Problem with SNOM Attendant phones - NeedHelp
Importance: High



 


Hi, 

One
of my biggest customer, with more than 300 hundred persons using asterisk, is
having bing problems with the SNOM phones used at the operator position
(Attendant Console). We currently have 3 operator attending incoming calls and
bypassing them to the rest of the company. 

We
did not find any alternative for SNOM to use as Attendant Console (phone for
the operator). We need Operators to have a keypad where the could observe the
most important extensions of the office, and be able to transfer a call very
fast (just pressing a button). 

Recently,
SNOM released a new model. The SNOM-360 phone, that was ideal for that.  However,
it only supports one keypad (around 30 extensions), althought previous models
were able to support several of them. SNOM tolds us they are working on this,
but they would not have a solution until end of the year (at least). We had pay
in advance for the models a lot of time ago, but it seems there is a lot of
problems with SNOM stock, and they did not got units to server us. Finally, and
due a hard preassure, SNOM sent us directly 3 used phones they got in their
lab. We received the password of only 2 of them. 

A
week after being in production, we started with a lot of problems with them.
Such as phones reboot themselfes each 10 minutes. We tryed to upload a more
recent firmware, but one of them got broken.  We have NOW in production
just on phone, that still reboots each 10 minutes.  Problem is on the SNOM
phone, since with all the phones we have connected to Asterisk everything goes
ok. 

Customer
is unable to attend all theirs calls, and is really upset.  I need to
react very fast. 

Trying
to anticipate to this, one of our developers did a program for the Operator.
That program used the Asterisk Manager API of asterisk in order to check
events, and allow the operator to use a program to chek extension status and
forward the call to any extension very fast. However, it seems that Asterisk
becomes unstable when has a lot of traffic and your are using one of those programas,
and having that program installed caused a lot of crashes (we think Asterisk
Manager API is not very stable, so far).  We do not like FOP because it is
complicated to use for an untrained operator, that could easily lost calls, and
do incorrect operations. We need a more powerfull software for this.


I
saw that 3COM has also attendace phones with a lot of extensions, but they do
not distribute yet that phone on Europe.  I
need a very fast and urgent solution for my operators, and have a phone for them
asap. 

Could
any one, please, suggest me any solution ? Any one could serve me really fast a
Attendance Phone, with a lot of extensions, for using on Asterisk to deal with
this problem? (please, no more SNOMs, customer does not want further to see them).


Since
this problem is really urgent I will appreciate if any one having any proposal
could contact directly with me at andres at europesip.com. Any suggestions of
specific phones with keypads (to be used as Attendance Phones) or programs we
could use for the operator, will help a lot. 

kind
Regards, 

Andres











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[Asterisk-biz] Serious Problem with SNOM Attendant phones - Need Help

[asterisk]

Hi,

One of my biggest customer, with more
than 300 hundred persons using asterisk, is having bing problems with the
SNOM phones used at the operator position (Attendant Console). We currently
have 3 operator attending incoming calls and bypassing them to the rest
of the company.

We did not find any alternative for
SNOM to use as Attendant Console (phone for the operator). We need Operators
to have a keypad where the could observe the most important extensions
of the office, and be able to transfer a call very fast (just pressing
a button).

Recently, SNOM released a new model.
The SNOM-360 phone, that was ideal for that.  However, it only supports
one keypad (around 30 extensions), althought previous models were able
to support several of them. SNOM tolds us they are working on this, but
they would not have a solution until end of the year (at least). We had
pay in advance for the models a lot of time ago, but it seems there is
a lot of problems with SNOM stock, and they did not got units to server
us. Finally, and due a hard preassure, SNOM sent us directly 3 used phones
they got in their lab. We received the password of only 2 of them.

A week after being in production, we
started with a lot of problems with them. Such as phones reboot themselfes
each 10 minutes. We tryed to upload a more recent firmware, but one of
them got broken.  We have NOW in production just on phone, that still
reboots each 10 minutes.  Problem is on the SNOM phone, since with
all the phones we have connected to Asterisk everything goes ok.

Customer is unable to attend all theirs
calls, and is really upset.  I need to react very fast.

Trying to anticipate to this, one of
our developers did a program for the Operator. That program used the Asterisk
Manager API of asterisk in order to check events, and allow the operator
to use a program to chek extension status and forward the call to any extension
very fast. However, it seems that Asterisk becomes unstable when has a
lot of traffic and your are using one of those programas, and having that
program installed caused a lot of crashes (we think Asterisk Manager API
is not very stable, so far).  We do not like FOP because it is complicated
to use for an untrained operator, that could easily lost calls, and do
incorrect operations. We need a more powerfull software for this.

I saw that 3COM has also attendace phones
with a lot of extensions, but they do not distribute yet that phone on
Europe.  I need a very fast and urgent solution for my operators,
and have a phone for them asap.

Could any one, please, suggest me any
solution ? Any one could serve me really fast a Attendance Phone, with
a lot of extensions, for using on Asterisk to deal with this problem? (please,
no more SNOMs, customer does not want further to see them).

Since this problem is really urgent
I will appreciate if any one having any proposal could contact directly
with me at andres at europesip.com. Any suggestions of specific phones
with keypads (to be used as Attendance Phones) or programs we could use
for the operator, will help a lot.

kind Regards,

Andres


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Re: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition

[M. MAINGAULT]

Hi,
i'm interested in your products. Please send me more details on my email.

Best regards
- Original Message - 
From: "Tom Hayden" <[EMAIL PROTECTED]>
To: "Commercial and Business-Oriented Asterisk Discussion" 


Sent: Wednesday, September 28, 2005 12:52 PM
Subject: Re: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition



Wow - this could be a really valuable peice of hardware, especially
someone looking for redundancy. Anyone got any specs?

--
Tom

On 9/28/05, Matt <[EMAIL PROTECTED]> wrote:

we are interested in your card, please send us details, etc.

Best Regards

Matt

High Performance Gigabit Clustering Appliance
http://www.xgforce.com/loadbalancer.html

eClustering Service
http://www.xgforce.com/eService.html

Gigabit 3U Tera Servers
http://www.xgforce.com/teraserver.html

Gigabit 2U Servers
http://www.xgforce.com/server2u.html

Gigabit 1U Servers
http://www.xgforce.com/server1u.html

__

- Original Message -
From: "Matteo Brancaleoni" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, September 28, 2005 12:50 AM
Subject: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition


> Hi all.
>
> I'm proud to announce that at SMAU 2005
> (the biggest IT expo in italy) we'll present
> a new pci card for the Asterisk PBX.
>
> This card will be a dual/quad GSM voice card,
> in order to have "mobile voice lines" without any
> external gateway in Asterisk.
> SMS also supported natively on the card
> (and an Asterisk application will be able to send/receive
> SMS with the card).
>
> Currently the card is compatible with gsm networks
> in EU and America (900/1800/1900 Mhz), and on
> first months of next year also for Brazil and Japan.
>
> For further info, come and visit us @ SMAU 2005
> in Milan, Italy, from 19 to 23 October.
> Hall 12 Booth H22.
>
> Also a mailbox [EMAIL PROTECTED] is open for requests,
> but please understand that full informations
> will be available only after the exhibition.
>
> --
> Come to visit us @ SMAU 2005
> >From 19 to 23 october
> Hall 12 Booth H22
>
> Matteo Brancaleoni
> System Administrator
> Tel  +39.02.70633354
> Sip  [EMAIL PROTECTED]
> Iax2 [EMAIL PROTECTED]
> ___
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>
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--
Tom
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Re: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition

[Tom Hayden]

Wow - this could be a really valuable peice of hardware, especially
someone looking for redundancy. Anyone got any specs?

--
Tom

On 9/28/05, Matt <[EMAIL PROTECTED]> wrote:
> we are interested in your card, please send us details, etc.
>
> Best Regards
>
> Matt
> 
> High Performance Gigabit Clustering Appliance
> http://www.xgforce.com/loadbalancer.html
>
> eClustering Service
> http://www.xgforce.com/eService.html
>
> Gigabit 3U Tera Servers
> http://www.xgforce.com/teraserver.html
>
> Gigabit 2U Servers
> http://www.xgforce.com/server2u.html
>
> Gigabit 1U Servers
> http://www.xgforce.com/server1u.html
>
> __
>
> - Original Message -
> From: "Matteo Brancaleoni" <[EMAIL PROTECTED]>
> To: 
> Sent: Wednesday, September 28, 2005 12:50 AM
> Subject: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition
>
>
> > Hi all.
> >
> > I'm proud to announce that at SMAU 2005
> > (the biggest IT expo in italy) we'll present
> > a new pci card for the Asterisk PBX.
> >
> > This card will be a dual/quad GSM voice card,
> > in order to have "mobile voice lines" without any
> > external gateway in Asterisk.
> > SMS also supported natively on the card
> > (and an Asterisk application will be able to send/receive
> > SMS with the card).
> >
> > Currently the card is compatible with gsm networks
> > in EU and America (900/1800/1900 Mhz), and on
> > first months of next year also for Brazil and Japan.
> >
> > For further info, come and visit us @ SMAU 2005
> > in Milan, Italy, from 19 to 23 October.
> > Hall 12 Booth H22.
> >
> > Also a mailbox [EMAIL PROTECTED] is open for requests,
> > but please understand that full informations
> > will be available only after the exhibition.
> >
> > --
> > Come to visit us @ SMAU 2005
> > >From 19 to 23 october
> > Hall 12 Booth H22
> >
> > Matteo Brancaleoni
> > System Administrator
> > Tel  +39.02.70633354
> > Sip  [EMAIL PROTECTED]
> > Iax2 [EMAIL PROTECTED]
> > ___
> > Asterisk-Biz mailing list
> > Asterisk-Biz@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-biz
> >
> ___
> Asterisk-Biz mailing list
> Asterisk-Biz@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz
>


--
Tom
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Re: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition

[Matt]

we are interested in your card, please send us details, etc.

Best Regards

Matt

High Performance Gigabit Clustering Appliance
http://www.xgforce.com/loadbalancer.html

eClustering Service
http://www.xgforce.com/eService.html

Gigabit 3U Tera Servers
http://www.xgforce.com/teraserver.html

Gigabit 2U Servers
http://www.xgforce.com/server2u.html

Gigabit 1U Servers
http://www.xgforce.com/server1u.html

__

- Original Message - 
From: "Matteo Brancaleoni" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, September 28, 2005 12:50 AM
Subject: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition


> Hi all.
> 
> I'm proud to announce that at SMAU 2005 
> (the biggest IT expo in italy) we'll present
> a new pci card for the Asterisk PBX.
> 
> This card will be a dual/quad GSM voice card,
> in order to have "mobile voice lines" without any
> external gateway in Asterisk.
> SMS also supported natively on the card
> (and an Asterisk application will be able to send/receive
> SMS with the card).
> 
> Currently the card is compatible with gsm networks
> in EU and America (900/1800/1900 Mhz), and on
> first months of next year also for Brazil and Japan.
> 
> For further info, come and visit us @ SMAU 2005
> in Milan, Italy, from 19 to 23 October.
> Hall 12 Booth H22.
> 
> Also a mailbox [EMAIL PROTECTED] is open for requests,
> but please understand that full informations
> will be available only after the exhibition.
> 
> -- 
> Come to visit us @ SMAU 2005
> >From 19 to 23 october 
> Hall 12 Booth H22
> 
> Matteo Brancaleoni
> System Administrator
> Tel  +39.02.70633354
> Sip  [EMAIL PROTECTED]
> Iax2 [EMAIL PROTECTED]
> ___
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> Asterisk-Biz@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-biz
> 
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RE: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition

[Mark Armstrong]

2N watchout, you may want to review this Matt 


Regards
 
Mark 

ACC Australia
Phone: 02 9489 0544
Fax: 02 9489 0522
Mobile: 0401 286 686
 
www.accaustralia.com.au
 

"The reality of VoIP and converged networks is simply put: They are here,
and the companies that implement them in an effective way will be the ones
to benefit first from the new services and cost savings they enable. The
companies that integrate VoIP technology will enjoy significant competitive
advantages."

 
___NOTICE
This electronic mail transmission contains confidential information intended
only for the person(s) named. Any use, distribution, copying or disclosure
by any other person is strictly prohibited. If you received this
transmission in error, please notify the sender by reply e-mail and then
destroy the message. Opinions, conclusions, and other information in this
message that do not relate to the official business of ACC Australia shall
be understood to be neither given nor endorsed by the Company. When
addressed to ACC clients, any information contained in this e-mail is
subject to the terms and conditions in the governing client contract. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matteo
Brancaleoni
Sent: Wednesday, 28 September 2005 5:50 PM
To: asterisk-biz@lists.digium.com
Subject: [Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition

Hi all.

I'm proud to announce that at SMAU 2005 (the biggest IT expo in italy) we'll
present a new pci card for the Asterisk PBX.

This card will be a dual/quad GSM voice card, in order to have "mobile voice
lines" without any external gateway in Asterisk.
SMS also supported natively on the card
(and an Asterisk application will be able to send/receive SMS with the
card).

Currently the card is compatible with gsm networks in EU and America
(900/1800/1900 Mhz), and on first months of next year also for Brazil and
Japan.

For further info, come and visit us @ SMAU 2005 in Milan, Italy, from 19 to
23 October.
Hall 12 Booth H22.

Also a mailbox [EMAIL PROTECTED] is open for requests, but please understand
that full informations will be available only after the exhibition.

--
Come to visit us @ SMAU 2005
>From 19 to 23 october
Hall 12 Booth H22

Matteo Brancaleoni
System Administrator
Tel  +39.02.70633354
Sip  [EMAIL PROTECTED]
Iax2 [EMAIL PROTECTED]
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[Asterisk-biz] Asterisk Quad Gsm card at SMAU exhibition

[Matteo Brancaleoni]

Hi all.

I'm proud to announce that at SMAU 2005 
(the biggest IT expo in italy) we'll present
a new pci card for the Asterisk PBX.

This card will be a dual/quad GSM voice card,
in order to have "mobile voice lines" without any
external gateway in Asterisk.
SMS also supported natively on the card
(and an Asterisk application will be able to send/receive
SMS with the card).

Currently the card is compatible with gsm networks
in EU and America (900/1800/1900 Mhz), and on
first months of next year also for Brazil and Japan.

For further info, come and visit us @ SMAU 2005
in Milan, Italy, from 19 to 23 October.
Hall 12 Booth H22.

Also a mailbox [EMAIL PROTECTED] is open for requests,
but please understand that full informations
will be available only after the exhibition.

-- 
Come to visit us @ SMAU 2005
>From 19 to 23 october 
Hall 12 Booth H22

Matteo Brancaleoni
System Administrator
Tel  +39.02.70633354
Sip  [EMAIL PROTECTED]
Iax2 [EMAIL PROTECTED]
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RE: [Asterisk-biz] voip.

[m fayyaz]

Hi!
We are based in Lahore, Pakistan. We have have different routes like Pakistan,India, BD Cell and Saudia Arabia which are good in quality as well as in rate. 
We provide 24X7 support.
REgards.Fayyaz[EMAIL PROTECTED]


From: "::Wholesale-Telecom:.:" <[EMAIL PROTECTED]>Reply-To: Commercial and Business-Oriented Asterisk DiscussionTo: Subject: [Asterisk-biz] voip.Date: Tue, 27 Sep 2005 08:58:34 -0600






Good morning, We are looking for direct wholesale routes world wide.
Please respond to [EMAIL PROTECTED]
 
 



















We Sell You What You Need, Not What We Have


 






Kevin Flake 

Wholesale-Telecom 


[EMAIL PROTECTED] 





tel: fax: mobile: 

303-471-2550 ext.3575720-528-8058303-332-2122 




 











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