Brandon,
What it sounds like you are looking at as far as having the phones
register to the system and then have users login to a phone should be
possible, I have not tried. I would suspect that you could build a
dial plan menu to prompt the caller for their credentials and then
take the phone's identification and add an entry to a database for
that phone and the caller id, extension, voice mail box etc.. You
could then use the realtime engine to query the table for the
information when an extension is dialed. Doing the login through a
sort of IVR would make it hardware independent.
One note on the QOS, You might be right about it being okay with only
3 calls at a time, but I would offer this example. We us a MPLS
network, in which most site have 384k and our main sites have 1.5mb.
When I first placed a test call over the link using gsm from a 384
site to the main site, they had no issues, but I had terrible problems
with audio being dropped or delayed and playing over top itself. I
implemented QOS because the connection is not only for voice, so I
could at least give priority to my audio and set aside bandwidth for
it.
On your time frame, it is hard to say, your users and existing
hardware and training all factor into it. Having done asterisk systems
before, I have deployed small sites, like 2-5 people in very short
time frames, typically a few days building the system off site and
testing then a week or less on site dealing with wiring, setup/testing
and training. On a site of your size I would almost consider spending
a couple weeks on site.
You mentioned trixbox, I started with [EMAIL PROTECTED] myself and must say it
was a
great thrill to place a call between two softphones after a hour or
so. But what I eventually realized was that if I had to troubleshoot
the dialplan I was going to be lost in macros and AGI. I started out
writing my configs and then using svn repositories for each site and
copying in a base config for each new site. Worked great for a while
and I knew the dialplan inside out. I'm now moving more and more to
realtime and database storage for config files and dialplan sections
which make managing multiple sites config files much easier. Also the
use of Dundi in [EMAIL PROTECTED] was not through the GUI, and I understood it
much
better once I began using the files. The bottom line: in my experience
[EMAIL PROTECTED]/Trixbox/freepbx are great ways to get your feet wet and are a
proof of concept and even great for a basic system, but for what you
are wanting to do and what I did. Asterisk is the only way to go. If
your worried about not knowing enough, goto a bootcamp or some other
training. If you want ease administration for several IT people, then
you could look at some of the web interfaces that connect and edit the
conf files.
Hope that helps
On 3/13/07, Brandon Comouche <[EMAIL PROTECTED]> wrote:
For startes I will keep it on the list and we can discuss some major
concepts, and I will possibly make some contact off list later for the
nitty-gritty :)
In-reply to Steve:
I did have a look at the "bicomsystems" product and it does appear to do
everything I am looking for. However, I have looked in to vendor systems
and have decided to go with an Asterisk system. Hench asking for
assistance on the Asterisk mailing list ;)
On the discussion at hand:
At this time I am not going to worry about the QoS with my T1 network
lines, I have been wondering what the quality will be like. I do not
plan to have more than maybe three calls on a line at peak times. But I
know that there will be more in the future. I am working with a total
employee base of around 30, and the remote offices have two to four
employees at a time, not a huge traffic demand.
What I am most curious about at this time is the methods used to move
from server to server. *Ideally* I would like to sit down at a phone,
enter my extension/password and have that phone ring as my extension.
Essentially, I would like a log in system on the phone. This presents me
with two issues: I have to make my phones allow simple logon as a SIP
device, and I need to get my credentials to move between Asterisk
servers. What methods have others used, or where should I look for more
information?
At this point I have two Polycom phones (430 and 501) for testing, they
seem to be talked about as very flexible. If they will not allow me to
add a "user friendly" login prompt, maybe I need to find alternatives
though. But this is the Asterisk list and I don't want to go too far off
topic, so the main concern is how I would synchronize my information
between asterisk servers.
One final topic on this message I would like to cover is time frame. I
am thinking maybe around 6 months to have at least a partial functioning
system up and tested. By partial I mean deployable with a basic
infrastructure feature set. I don't know if this is too little time or
too much time. My co-workers are excited about what Asterisk has to
offer. Any other thoughts on time frames?
P.S. I want to thank everyone who replied so quickly, surprised my
co-worker and I :D
--
Thanks,
Brandon Comouche
IT Administrator
Sno Falls Credit Union
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent: Monday, March 12, 2007 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: Seamless Multi Office Asterisk
Deployment
I'm more then happy to share my experiences with anyone, there is just
a lot to be said about the things Brandon is trying to accomplish.
Take the automatic fail over he mentioned, there are a number of ways
to do that and everyone has an opinion. I just want make myself
available to help other get from "playing" with Asterisk like I did to
really putting it to use so that people sit back and say wow, my
cisco/avaya/nortel can't do that.
On 3/12/07, Sean Bright <[EMAIL PROTECTED]> wrote:
> Why does everyone want to go off-list? Is this not information that
could
> benefit others?
>
>
> On 3/12/07, Bruce Reeves < [EMAIL PROTECTED]> wrote:
> > Brandon
> >
> > Your on the right track with what is can do. It will also be good to
> > look into what kind of QOS you can do on the T-1 connections between
> > offices. I have an 8 office setup similar to this and many of your
> > goals I have achieved and would be glad to offer ideas and such if
you
> > want to email me off list.
> >
> > On 3/12/07, Brandon Comouche <[EMAIL PROTECTED]> wrote:
> > >
> > >
> > >
> > >
> > > Hello
> > >
> > >
> > >
> > > I have a brief and a long question about a possible Asterisk
deployment
> I am
> > > planning.
> > >
> > >
> > >
> > > Long Story Short:
> > >
> > > I have four total offices, one main and three remote. All offices
are
> > > connected using dedicated network T1 lines creating one unified
network
> > > across offices. I would like to know if it is possible to set up
an
> Asterisk
> > > system with the following capabilities:
> > >
> > > - Branch Unification (I know this can be done)
> > >
> > > - Branch Independence (In case of T1 network Failure, PSTN line
failover
> at
> > > each branch)
> > >
> > > - Roaming Extensions (A user can go to any office and log in to a
phone
> -
> > > hopefully check voice mail as well)
> > >
> > > Basically, I am asking if Asterisk can be a system that will
seamlessly
> > > operate as one big system and handle failovers as well.
> > >
> > >
> > >
> > > After spending hours playing with Asterisk, reading voip-info.org,
and
> > > watching this list, it seems that Asterisk can handle anything. I
just
> would
> > > like re-assurance that I am not chasing a lost cause. A simple Yes
or No
> > > answer is acceptable to me. Below I have a long version of what I
am
> trying
> > > to do if anyone is in the mood to give me more pointers J
> > >
> > >
> > >
> > >
> > > Brandon
> > >
> > > (Long Version Follows)
> > >
> > >
> > >
> > > Long Story Version:
> > >
> > > Here is what I have to work with:
> > >
> > > - Four Offices (One main and three remote)
> > >
> > > - Dedicated T1 lines connecting three remote offices to one main
office
> (all
> > > connections made through the main office)
> > >
> > > - Will have a T1 Voice line at the main office
> > >
> > > - Three PSTN lines at each remote office
> > >
> > >
> > >
> > > Essentially what I would like to do is create a system comparable
to the
> > > ShoreTel ShoreGear product line (if you are familiar with it).
This
> system
> > > will seamlessly unite all offices as one and provide failover in
the
> case of
> > > line outage. It also allows users to roam from phone to phone
across
> offices
> > > seamlessly. It has many more features, but those are two main
features I
> am
> > > looking for. About 40 total phones will be deployed. To make it
even
> more
> > > difficult, I would like all user extensions to start the same
(i.e.
> across
> > > offices all extensions are 5### with no discernable pattern).
> > >
> > >
> > >
> > > Progress so far:
> > >
> > > At this time I have determined that I will need a server at each
office
> as
> > > well as a T1 card (TE110P) at the Main office and the four port
TDM PSTN
> > > cards at each remote office. I plan on using the Polycom IP 430 or
501
> > > (Undecided, 501 if required). I have been using TrixBox to this
point,
> would
> > > like to continue if possible. It appears that I will want to use
DunDi
> in
> > > some fashion to unite the branches.
> > >
> > >
> > >
> > > My main roadblock right now is trying to figure out how to get all
the
> > > information across the offices at the same time (extensions,
voicemail).
> I
> > > have successfully had two boxes communicate, but what I am looking
for
> is
> > > much more complex I feel. I have thought of synchronized MySQL
> databases,
> > > but I do not know if that will work the way I wish.
> > >
> > >
> > >
> > > If anyone reads this far ;) I am looking for suggestions for
routes I
> might
> > > consider or places I should/could look for more information. I am
> relatively
> > > new to Asterisk, but I am not afraid to get my hands dirty. If
something
> I
> > > said did not make any sense or if there is more information I
could
> provide,
> > > I am happy to help where I can. Thank you for your time and
assistance.
> > >
> > >
> > >
> > > Brandon Comouche
> > > An IT Guy
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
> >
> > --
> > Bruce
> > Nortex Networks
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> sean
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
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>
>
--
Bruce
Nortex Networks
_______________________________________________
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To UNSUBSCRIBE or update options visit:
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_______________________________________________
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Bruce
Nortex Networks
_______________________________________________
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