Re: [asterisk-users] High Port Count ATA

2007-06-01 Thread Natambu Obleton

i have deployed the audiocodes mp-124? with 14 lines active lines and
it ugly to configure, but works well once setup. They do make it easy
if you have a set of contiguous number to apply to the ports in order
though.

On 5/31/07, Jay R. Ashworth [EMAIL PROTECTED] wrote:

On Thu, May 31, 2007 at 01:22:06PM -0700, Douglas Garstang wrote:
I'm trying to find a high port count ATA device. We have a lot (up to 110)
analog devices that we need to bridge to IP. Rather than go out and buy 110
ATA's, it would make more sense to buy a single chassis type device with
some number of ports and blades. Anyone know if such a device exists?

Sure; lots of them, in port counts up to 48.  They're not cheap.
Expect to pay about $105 a port or so, on average.

Start here:

http://www.voipsupply.com/index.php?cPath=94_286

I was looking at the Audiocodes, about 6 months ago when I was prepping
a project, but I haven't had hands-on.  Iv you *really* need a lot of
ports, check out the Vegastream, but, again, no testimony here.

Keep in mind though, that it can be a good thing to have a warm spare,
so high port counts aren't *always* the best answer.

Cheers,
-- jra
--
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274
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Re: [asterisk-users] Simultaneous Capacity

2007-05-15 Thread Natambu Obleton

What about using a Lucent TNT and an asterisk box. Is the limit the
TDM--G.711 conversion?

On 5/14/07, Alex Balashov [EMAIL PROTECTED] wrote:

On Mon, 14 May 2007, Kapil Dhawan said something to this effect:

 I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is
 it feasible.

   In truth, it is very unlikely.

   How are you planning to pick up the PRIs, anyway?  3 quad-span T1 cards?

--
Alex Balashov   [EMAIL PROTECTED]
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Re: [asterisk-users] Re: Open CallerID Database?

2007-02-21 Thread Natambu Obleton

Just out of interest: From former posts I understood that there is a
CALLERID service in US (for an extra fee, I assume) that gives both
number _and_ name of the caller...? I am aware of the fact that e.g.
EuroISDN lines can transmit alphanumeric callerid (and in fact I already
use that on an ISDN phone here that connects to an Asterisk - showing a
few special names like wakeup call). Not for names yet, as I was too
lazy to implement that. Does that also work over analogue lines?


ISDN is america does the same thing, but in Canada( which could be
like europe) the callerid is transfered on the SS7 ISUP message, so
the owner of the number can provide their own callerid. In America
though, this isn't the case. You can not transfer the callerid over
the ISUP message, it must be looked up using TCAP(verisign) or IP
servce(targus).

It's not really getting the CallerID info to the local switch, but
getting it updates in this database in the sky that we are forced to
use. :)


On 2/20/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:

Am Dienstag, den 20.02.2007, 16:33 -0700 schrieb Natambu Obleton:
 I would guess that registration would be by the telco for the blocks
 just like with reverse dns today, so then each telco would have a
 local server to manage their 'reverse' cnam lookup and the people
 in charge would be NANPA, just like how ARIN is regulated today.
 Although who owns the root namservers.. I wonder if ARIN and RIPE
 share ownership of them?

 Although now that i Think about it dns wouldn't work because want to
 deligate .. XXX-XXX-XYYY and XXX-XXX-XXYY and then there is single
 numbers. For that right now I do weird PTR CNAME to A record thing for
 single reverse dns. This would be little larger... ohh shit.. LNP. So
 now Qwest would need to deligate a single CNAM to me and crap..naw
 this will never work.

I do not get your point here.

Take ENUM. A made-up phone number like +49 (228) 91234567 would be found
as
7.6.5.4.3.2.1.9.8.2.2.9.4.enum-something

This means, you delegate the 2.1.9.8.2.2.9.4.enum-something to the telco
that owns the 912 block in Bonn, Germany.

Number portability screws this, as even single numbers out of a
contiguous range of MSNs on a ISDN line can be moved over to another
provider, with the others staying with the old provider. They would have
to play well together, and that will fubar for sure (they even
sometimes block the DSL frequencies on lines when a customer moves to
another company, with unblocking taking a 14-day security period,
because they can - there is no technical block, but the DSLAM
manipulation database software will mark your DSL line as blocked,
making moving to another provider a pain with up to 6 weeks without
internet).

In Germany we have an agency that manages all phone book data, and
(nearly?) all the 411 type services (called 118xx here) buy data there.
How they manage their data internally is none of my business (although I
would guess it's something like MSSQL with an Access frontend, thinking
about the Deutsche Telekom ;-).

They will no way accept DNS-type queries for free. Some months ago,
there was a heise.de (German IT newsticker) article about them charging
enourmous sums, paying the real data storage and administration costs
back by about factor 5 or so. Well-paying businesses rarely give away
their business turnpoint.

Any non-official system will suffer even worse inaccuracy than the
providers' own and managed system (as someone else already wrote). Their
data is quite bad enough. This relates, of course, to the fact that they
may only reverse-lookup numbers to find names if the customer
explicitely allowed them to do it, on the line rental agreement. There
are usually several checkboxes, allowing you to get listed in
phonebook, get listed in digital listings, and get listed for
reverse lookup.

For those who allow it there is a free web-frontend to reverse-lookup
numbers, which is a pain to script-access, but it is possible. It
suffers from problems with DIDs, as for example a shop might have the
number 94144-0, and assigned the numbers up to 94144-29. If you try to
lookup 9414488 (which might be a private person's analogue line, and
this is absolutely valid in the German numbering system) it will return
the wrong entry because the logic in their webinterface always assumes
that 94144-0 means all numbers starting 94144- belong to that line.
_That_ really sucks - you think business and then it's a friend calling
for private talk.

Just out of interest: From former posts I understood that there is a
CALLERID service in US (for an extra fee, I assume) that gives both
number _and_ name of the caller...? I am aware of the fact that e.g.
EuroISDN lines can transmit alphanumeric callerid (and in fact I already
use that on an ISDN phone here that connects to an Asterisk - showing a
few special names like wakeup call). Not for names yet, as I was too
lazy to implement that. Does that also work over analogue lines?

BR
Anselm

Re: [asterisk-users] Open CallerID Database?

2007-02-20 Thread Natambu Obleton

Why not make it like DNS and have each provider have their lookups
deligated to a local server and then each ISP will run a caching
server that will use a serial number system to get updates.. just like
DNS.

I know there are lot more DNS lookups then CNAM lookups per hour...
isn't there? :)

On 2/20/07, Mike Lynchfield [EMAIL PROTECTED] wrote:

With all other things said.. you might want a professional service for this
like  targusinfo.com

Maintaining and running an operation like a cname web lookup thing is REALLY
high overhead in terms of web traffic etc

What happens when you get 30 ITSP/clients pulling 1000 calls each or 10
calls each per day..

that can easily go up to 1 mill requests per day ,

How will you pay for the bandwith/hardware/failover/load balance etc
hardware for all this ?

or if you are going to charge then why reinvent the wheel.

targusinfo.com is what we would use..

Cname lookup is a really controversial matter , no one wants to absorb the
costs , that is why some TELCOS charge 4.95  for callerid ( its basically
the lookup service they are paying for) ..

CNAME lookups is also not mandatory for TELCOS so some do it some don't ,
but FREE cname is just not going to happen untill some one has a Return on
Investment strategy for this..


Take a look at Free 800 systems that went down , Any venture needs a capital
source of income..

my 0.02



On 2/20/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote:





 Hey Guys,
 I'm glad to see this ignited some discussion.

 I definitely understand there's some legal implications involved, both on
a privacy level, and fraud prevention. Obviously an end-user (ie: the person
controlling a listing) has to consent to some sort of release resolving the
privacy concerns. I'm somewhat aware of the legal implications involved with
storing such personally identifiable information (or whatever the legal term
is) and have a concern in making sure such issues are resolved.

 In reality, how is it efficient for every provider to be running their own
database? In my mind, this leaves the horribly evident inaccuracies, and
even efficiency issues. Thank God these accuracies aren't integral to the
operations of telephony systems.



 I do understand there is a price to pay for such infrastructure, and I
believe that it's obvious the telephony world is riddled with racketeering,
price gouging ventures, including companies that charge nearly a $0.01 for a
lookup. I realize the following analogy is poor, but in mind this is as
close as a internet search engine charging for a basic search query. Infact
a basic internet query is much more complex, much more costly (ie: the
infrastructure of said systems), and yet self-subsidizing.


 And to the poster who suggested that I was implying scrapping the results
from 411.com, this is definitely not even a remote idea in my mind at all.
The basis for my idea was a open, moderated, database that was user
controlled and self-subsidized.



 I know this is way off topic, but I really feel that the telecom industry
as a whole, and I'm sure I'm not the only one with this belief, is horribly
bloated, running on business models that are clearly 30 years outdated. It
is 2007, and with the help of the internet, the exchange of information,
these telcos now have real, global competition, and real issues to deal
with.

 Anyways guys, I'm curious to hear your thoughts.




 --
 Robert Norton
 SophMedia LLC Operations Manager
 Cell: 480-234-4312 Office: 480-626-5449 (x300)
 P.O. Box 7755 Tempe, AZ 85281
 http://www.XStreamHost.com - Web Hosting
 http://www.SophMedia.com - Consulting  Web Development






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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Re: Open CallerID Database?

2007-02-20 Thread Natambu Obleton

So how does this start? I mean it wouldn't be hard to modify dns
server to use 3/3/4 format ip address... or it would need to be
3/3/3/4 for international right and someone wrote a module for
asterisk look up that way and then I took my SS7 connection and setup
a GTT gateway to a server so that real telco's could query it over SS7
and use my other CNAM provider as backup so that people could make me
their main connection, but alas...

most cnam providers don't allow you to resale lookups. Time to move
on, but it would be kewl to start a revolution. :)

On 2/20/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:

Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield:
 Well caching is the way to go., bu then again most of the current
 solutions have this problem.

 John smit has a DID.. 514 555 1234 and closes account.. did sleeps for
 3 months and new client Jane doe takes it..

 Now how long should caching be ? this is a big problem ATM because
 some cache for 1 year others 1 day , they don't want to tell how long
 nor provider an API update method.

Coming back to the DNS example, there are certain timeouts. I have to
admit I cannot tell how exactly the timeout values work together, but
you _can_ set an absolute timeout after which any cached data (counted
from the moment of retrieval) is marked obsolete and a subsequent query
occurs. If you set something in the 2-week-range (which may or may not
be what many people use in DNS) you can be pretty sure that freshly
assigned numbers do not have dangling cache records, assuming the 3
months gap before assigning the same number again.

Assuming one could add an additional TXT record to enum, say

name.0.6.0.7.x.x.x.enum.info. TXT Hoffmeister, Anselm Martin

this would pretty much do the trick. I have no idea wether any standard
describes name resolution via enum.

The other way around would be more tricky btw., with all those John
Smith around ;)

BR
Anselm

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Re: [asterisk-users] Re: Open CallerID Database?

2007-02-20 Thread Natambu Obleton

I would guess that registration would be by the telco for the blocks
just like with reverse dns today, so then each telco would have a
local server to manage their 'reverse' cnam lookup and the people
in charge would be NANPA, just like how ARIN is regulated today.
Although who owns the root namservers.. I wonder if ARIN and RIPE
share ownership of them?

Although now that i Think about it dns wouldn't work because want to
deligate .. XXX-XXX-XYYY and XXX-XXX-XXYY and then there is single
numbers. For that right now I do weird PTR CNAME to A record thing for
single reverse dns. This would be little larger... ohh shit.. LNP. So
now Qwest would need to deligate a single CNAM to me and crap..naw
this will never work.

On 2/20/07, Jon Pounder [EMAIL PROTECTED] wrote:

Quoting Natambu Obleton [EMAIL PROTECTED]:

 So how does this start? I mean it wouldn't be hard to modify dns
 server to use 3/3/4 format ip address... or it would need to be
 3/3/3/4 for international right and someone wrote a module for
 asterisk look up that way and then I took my SS7 connection and setup
 a GTT gateway to a server so that real telco's could query it over SS7
 and use my other CNAM provider as backup so that people could make me
 their main connection, but alas...

I think you have the format of the address space reversed conceptually but the
idea would work - the countrycode, area code etc are like the .com in dns and
then exchange like the 2ld, and number like the hostname.

the only modification would be to set different root servers for this sort of
parallel system, and have someone actually in charge to delegate the
subdomains
etc.

registering a domain name has cost for the admin part of it, but is someone
really going to pay to register a phone number they already pay for ? (the
administration has to be paid for somehow)

I am not trying to criticize, but just pointing out the realities of making it
work.



 most cnam providers don't allow you to resale lookups. Time to move
 on, but it would be kewl to start a revolution. :)

 On 2/20/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
 Am Dienstag, den 20.02.2007, 14:54 -0500 schrieb Mike Lynchfield:
  Well caching is the way to go., bu then again most of the current
  solutions have this problem.
 
  John smit has a DID.. 514 555 1234 and closes account.. did sleeps for
  3 months and new client Jane doe takes it..
 
  Now how long should caching be ? this is a big problem ATM because
  some cache for 1 year others 1 day , they don't want to tell how long
  nor provider an API update method.

 Coming back to the DNS example, there are certain timeouts. I have to
 admit I cannot tell how exactly the timeout values work together, but
 you _can_ set an absolute timeout after which any cached data (counted
 from the moment of retrieval) is marked obsolete and a subsequent query
 occurs. If you set something in the 2-week-range (which may or may not
 be what many people use in DNS) you can be pretty sure that freshly
 assigned numbers do not have dangling cache records, assuming the 3
 months gap before assigning the same number again.

 Assuming one could add an additional TXT record to enum, say

 name.0.6.0.7.x.x.x.enum.info. TXT Hoffmeister, Anselm Martin

 this would pretty much do the trick. I have no idea wether any standard
 describes name resolution via enum.

 The other way around would be more tricky btw., with all those John
 Smith around ;)

 BR
 Anselm

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Jon Pounder

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Thorold, Ontario, Canada

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RE: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Natambu Obleton
Yes, but they require a software upgrade. I am using them for plain ole
dialup, but because of that feature set I figured someone on this list would
know how. I find a lot of people online saying to serial into them and reset
them, but no detail on when to send break or what dip switches to set or
anything. :(


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Wednesday, October 04, 2006 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TNT Max Password reset

On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
Anyone have happen know how to reset the password on a TNT Max? Thanks.

Does your asking here suggest that the the MAX's can do, say, voice
gateway service?  Protocols?  Codecs?

Cheers,
-- jra
-- 
Jay R. Ashworth
[EMAIL PROTECTED]
Designer  Baylink RFC
2100
Ashworth  AssociatesThe Things I Think'87
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647
1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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RE: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Natambu Obleton
maxrouter.com - You are on the page : Voice over IP
http://lnk4.us/VQ02


MAX Routers from Ascend / Lucent for VOIP ?

On these pages we are presenting the use and knowledge of used Ascend/Lucent
and used Compaq server products in the foreground. However, there are more
vendors with excellent equipment for gateways and servers.

As we know, the Ascend MAX 3000 and MAX 6000 and the MAX TNT are able to
handle VOIP connections. These units are able to run multi channel (multi
port) DSP based cards for coding and compression of natural speech.

The MAX 3000 ist rarely sold on the market but the MAX 6000 and MAX TNT are
out in large quantities. So let us focus on these units.

MAX 1800, MAX 2000 and MAX 4000 are not expandable to run VOIP DSP boards.


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Thursday, October 05, 2006 8:49 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TNT Max Password reset

On Thu, Oct 05, 2006 at 10:28:21AM +0100, Steve Kennedy wrote:
 On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
  On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
  Anyone have happen know how to reset the password on a TNT Max?
Thanks.
  Does your asking here suggest that the the MAX's can do, say, voice
  gateway service?  Protocols?  Codecs?
 
 Ascent TNT's with the right software and hardware can do SIP, E1
 termination/origination, and all sorts of codecs.

To clarify, since I have an older Max: That's only the TNTs?

I'm pre-speccing a project I've mentioned a couple times before; 390
port hospitality PBX, and I'm always looking for good FXS gateways...

Cheers,
-- jra
-- 
Jay R. Ashworth
[EMAIL PROTECTED]
Designer  Baylink RFC
2100
Ashworth  AssociatesThe Things I Think'87
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647
1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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[asterisk-users] TNT Max Password reset

2006-10-04 Thread Natambu Obleton








Anyone have happen know how to reset the password on a TNT Max?
Thanks.





Natambu Obleton

Network Engineer

FastTrack Communications

[EMAIL PROTECTED]

(970) 247-3366 office

(970) 247-2426 fax










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[asterisk-users] ANI and Meetme...

2006-09-18 Thread Natambu Obleton








Ok. First question is how to make it say my number back.

Like if you call extension 1000 from extension 1001, I want
it to say Number is 1,0,0,1 like an ANI number? Help.





Also I want to setup a meetme conference so that it asks Enter
conference number then execute meetme($entered_number) 





I feel dumb asking because these sound like they should be
so easy, but I cant find any help with this. Thanks.





Natambu Obleton

Network Engineer

FastTrack Communications

[EMAIL PROTECTED]

(970) 247-3366 office

(970) 247-2426 fax










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RE: [asterisk-users] Fedora

2006-09-18 Thread Natambu Obleton
Centos is a free version of RHEL.

Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Monday, September 18, 2006 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fedora

Yes.  We use fedora as a test system before moving it onto RHEL.

On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote:
 Hi list;
 
 Does asterisk work with fedora because redhat
 enterprise is licensed and costly.
 
 Regards
 Bilal
 
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-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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RE: [asterisk-users] SER Dispatcher Load Balance How-To?

2006-08-30 Thread Natambu Obleton
If you want a good explaination of SER and how to use it start here. 

http://siprouter.onsip.org/doc/gettingstarted/

They have GREAT pre-written configs and walk you through ever part of SER. I
was about scrap SER before I found these tutorials.


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, August 30, 2006 8:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, August 29, 2006 11:07 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?
 
 
 Well, it really depends on what he's using the asterisk 
 servers for.  If
 it's for voicemail or apps, he'll just have to make sure that certain
 apps land on certain servers, and voicemail can be distributed for
 various things.  If ser can do what I've heard/read it can do, it can
 handle all the basic call functions (i.e. forwarding) for plenty of
 calls.  Also, if the asterisk servers are just acting as 
 gateways (i.e.
 t1, e1, etc), then they will have no problem handling a load balanced
 configuration.

To do that, you'd need to use the avpops module in OpenSER. You think
Asterisk documentation is bad? Wait until you try and get that stuff to
work.
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RE: [asterisk-users] question about asterisk DB

2006-07-21 Thread Natambu Obleton
There is a nice one called RealTime. Works great and you can even get it to
store email in sql, but I haven't really tested it much.

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

http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of unplug
Sent: Friday, July 21, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question about asterisk DB

Hi,
  What is the mechanism of asterisk DB?  In folder /var/lib/asterisk,
there is a file called astdb.  Does it used for storing data of
asterisk DB?
thanks,unplug
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RE: [asterisk-users] A very lost newbie.

2006-07-20 Thread Natambu Obleton








http://revision3.com/systm/asterisk/



Nice video podcast and they even give you
the files they worked with on the show.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David R.
Sent: Thursday, July 20, 2006 4:40
PM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] A very
lost newbie.





Here's my situation. I'm a programmer at the ISP I work for and
my supervisors have seen fit for me to become an administrator for our new
ventures into the VoIP world. All of us are relatively new at this jazz,
though naturally my higher ups have more experience than I. We're toying
with VoIP a lot in the office, and I personally feel like I'm fairly far behind
them in knowledge of it all. 

I've done some studying the past day or two, and I grasp the basic principles
of VoIP, Asterisk, and SER/OpenSER, and about 5% of the hardware involved in
all of it (programmer; software guy). I'd like to catch up to them as
quickly as possible so I can be of most use. Here's what I have available
to me, and what I've done so far: 

AVAILABLE:
- A box with OpenBSD (my preferred OS for most production servers)
- Asterisk installed on the box.
- One IP Phone

DONE:
- Installed Asterisk.
- Written this e-mail. :)

I'm wondering where to begin. I have Asterisk installed but don't know
how to go about getting my IP phone to work with it. To start, I'd like
to make calls only internally on my home network (not even screwing with my
company's network yet; they're further along into configs than I want to get
into just yet) to a soft phone that Ubuntu Dapper Drake (my OS at home) comes
with. 

My question is this:

Where can I find good starter documentation(s) for my purposes?

Thank you,
David






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[asterisk-users] Current radius patches

2006-07-17 Thread Natambu Obleton








I would like to setup asterisk with Realtime and radius authentication,
but the radius patches are either outdated ( they support a version of asterisk
before realtime was mature ) or they dont patch right.



I tried this, but the version it is for is really old.

PortaOne Radius auth - voip-info.org

http://lnk4.us/pZfq



I tried this here

0005424: [branch] SIP peer authentication on an external
database (RADIUS - LDAP) - Mantis

http://lnk4.us/Ky7h





But those patches seem to line up with a Revision 36170,
which isnt in the SVN server. So before I be dumb and just apply the
patches by hand. Anyone know where I can get current patches for radius authentication
for asterisk 1.2.x? Thanks.






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